[Asterisk-Users] Expression GotoIf - bug or personal misunderstanding?
Hi, I am using 1.2.4 of asterisk. From the console: -- Executing GotoIf(Zap/29-1, 1 0?4:3) in new stack -- Goto (macro-stdexten,s-NOANSWER,4) In my understanding the expression (1 0) should be lead to 0, but in this case it leads to 1. Can anybody explain this to me? Much thanks ___ --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] ATA's and faxing
Garth, this is my sip-configuration for a fax machine at a AT386 ; SIP Accounts Analog devices like Faxmachines [analogdefaults](!) type=friend host=dynamic dtmfmode=info disallow=all allow=gsm allow=alaw allow=ulaw [222](analogdefaults) context=sip-ol callerid=Fax 222 username=222 secret=123 ; The ATA adapter itself is configured as follows: Fax Mode: (x) T.38 (Auto Detect)Pass-Through so, I don't even have configured Pass-Through Bernd Garth van Sittert wrote: I am using alaw and I have already enabled the pass through. Does alaw and ulaw work? I can fax out, but not receive faxes. Garth Johann Steinwendtner wrote: Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config. Hans Garth van Sittert schrieb: Hi All Is there any special configuration needed to send and receive faxes on an ATA device? I am using G711.a with a Grandstream Handytone 486. I can send faxes from a fax machine on the ATA, but receiving doesn't work. I get the fax signal, but it just doesn't continue. The LAN is used purely for VoIP traffic. Garth ___ --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 ___ --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] Help with Grandstream Handytone 386 together with Asterisk and a connected modem
Hello, we use Handytone 386 adapter together with the Asterisk PBX. Using normal analog phones together with the Handytone and Asterisk works fine. We also can connect a standard fax machine to the Handytone ATA adapter.Send and receive of faxes works fine. When we connect a standard analog modem to the Handytone adapter we can establish outgoing calls but when we try to call the modem we never get an answer or see any incoming call on the modem. What could be the problem? Thanks and regards, ___ --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] Interface card for Euro-ISDN (BRI)
we are using the beronet cards together with mISDN, works stable on system with digium and beronet we use bristuff John Jensen wrote: Hi, I'm looking for an interface card for termination of Euro-ISDN2 (BRI) lines. That is ISDN lines from the telco into my Asterisk box. Any recommendations, good/bad expiriences ? At present I'm looking at cards from BeroNet and Junghanns. Cheers, John Faroese Telecom ___ --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 ___ --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] MoH trouble with latest bristuff (0.3.0-PRE-1f)
Hi Karsten, I have the same problem. MusicOnHold sounds awful. The PRE-1e does not have this problem. I have two identical systems (hard-/software). One system has the problem the other does not. I thought i could be timing problem or interrupt conflict. But we could not find out the problem. Bernd Karsten Wemheuer wrote: Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but there was another issue, so I have to upgrade). Thanks in advance, Karsten ___ --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 ___ --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] MoH trouble with latest bristuff (0.3.0-PRE-1f)
Tzafrir Cohen wrote: On Tue, Jan 17, 2006 at 03:48:44PM +0100, Kib Eki wrote: If I use a rawplayer like this: #!/bin/sh while(true) do for name in $@; do cat $name ; done done BTW: 'while(true)' is is csh syntax that accidentally works in sh. In sh it spawns a subshell for the true. BTW:[2] can you think of a way to use the (bash-specific) $RANDOM to play a random file of [EMAIL PROTECTED] Sorry, but i am not that much shell expert. I stole the script from OrderlyQ ___ --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] Errors with bristuff-0.3.0-PRE-1e and asterisk cores
Hi, can anybody tell me what the errors mean and why my asterisk server falls from time to time. From time to time means several hours, not regularly. I also can provide a core if someone can debug? Thanks and regards Jan 11 14:34:59 NOTICE[13573] chan_zap.c: Hangup, did not find cref 83, tei 64 Jan 11 14:34:59 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:03 NOTICE[13573] chan_zap.c: Hangup, did not find cref 83, tei 64 Jan 11 14:35:03 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:20 NOTICE[13573] chan_zap.c: Hangup, did not find cref 84, tei 64 Jan 11 14:35:20 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:24 NOTICE[13573] chan_zap.c: Hangup, did not find cref 84, tei 64 Jan 11 14:35:24 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:44 WARNING[13573] chan_zap.c: Whoa, there's no owner, and we're having to fix up channel 22 to channel 23 Jan 11 14:37:44 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:37:54 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:04 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:14 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:24 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:34 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:44 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). Jan 11 14:38:54 WARNING[13568] chan_zap.c: 3 received SETUP message for call that is not a new call (retransmission). ___ --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] Problem with octo bri
Hi, I have similar problems. Did you find out what is the problem? We use TE205P and Octro Bri card. At the Octo Bri there ISDN Router connect which dial out over the TE205P card. Regards Miloš Kocbek wrote: Hi I have octo bri card connected to 4 telco lines and 4 alcatel PBX lines. After few hours of calling i get message Ring requested on unconfigured channel 255/255 span 1 this message occurs immediate when call get to asterisk and it is denied immediately. If i restart asterisk it is working ok for few hours. greetings mk ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --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] Problem with blind transfer and Polycom phones
Hi, we just set up an asterisk with 55 Polycom 500 IP phones. The blind transfer does not work. The way we try to blind transfer a call: 1. answer the call 2. press transfer 3. press blind softkey - the display shows Blind transfer to: and cursor is in the second line 4. enter the number - when we enter the second digit of the number the display jumps back to Hold: number view. It is reproducible. Attended transfer works. Any help is welcome! :-)) Regards, BK ___ --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] Problem with blind transfer and Polycom phones !! more info
I have to correct myself. The problem occurs only when we try dial numbers with 10 or 11 at the beginning. Kib Eki wrote: Hi, we just set up an asterisk with 55 Polycom 500 IP phones. The blind transfer does not work. The way we try to blind transfer a call: 1. answer the call 2. press transfer 3. press blind softkey- the display shows Blind transfer to: and cursor is in the second line 4. enter the number- when we enter the second digit of the number the display jumps back to Hold: number view. It is reproducible. Attended transfer works. Any help is welcome! :-)) Regards, BK ___ --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 ___ --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] HELP! Asterisk 1.2.1 stops immediately - voicemail problem?
Hi, our production system stops immediately when a caller hangs up without leaving a voicemail. This is the last output from the console: -- Playing 'vm-isunavail' (language 'de') -- Playing 'vm-intro' (language 'de') -- Playing 'beep' (language 'de') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: wav49, 0x821ba18 -- x=1, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: gsm, 0x823a0f0 -- x=2, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: wav, 0x823a3d8 -- User hung up What can be wrong? Thanks for any help!! BK ___ --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] HELP! Asterisk 1.2.1 stops immediately - voicemail problem? - SOLVED
we removed the settings for emailbody and emailsubject Kib Eki wrote: Hi, our production system stops immediately when a caller hangs up without leaving a voicemail. This is the last output from the console: -- Playing 'vm-isunavail' (language 'de') -- Playing 'vm-intro' (language 'de') -- Playing 'beep' (language 'de') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: wav49, 0x821ba18 -- x=1, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: gsm, 0x823a0f0 -- x=2, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: wav, 0x823a3d8 -- User hung up What can be wrong? Thanks for any help!! BK ___ --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 ___ --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] Feature: Attendet transfer with original caller ID
Hi, I know that this has been an issue in older threads, but again i want to know when that feature will be available in Asterisk. Could anybody tell what the current plans for this are? Thank you very much, BK ___ --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] Polycom 500 IP and problems with show hints
Hi,
why does asterisk always give the state Unabailable?
asterisk-er*CLI
-= Registered Asterisk Dial Plan Hints =-
31 : SIP/31exten = 31 State:Unavailable Watchers 0
23 : SIP/23exten = 23 State:Unavailable Watchers 0
22 : SIP/22exten = 22 State:Unavailable Watchers 0
21 : SIP/21exten = 21 State:Unavailable Watchers 0
- 4 hints registered
asterisk-er*CLI
The phones register with only one SIP-extension. We use SIP Version 1.5.3 for
the phones.
Part from my extension.conf:
exten = 21,hint,SIP/21
exten = 21,1,Macro(stdexten,21,SIP/21,${CALLERIDNUM},${CIDNAME})
Thanks for any help!
Regards,
BK
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Re: [Asterisk-Users] Polycom 500 IP and problems with show hints - solved
complete restart of asterisk
Kib Eki wrote:
Hi,
why does asterisk always give the state Unabailable?
asterisk-er*CLI
-= Registered Asterisk Dial Plan Hints =-
31 : SIP/31exten = 31 State:Unavailable
Watchers 0
23 : SIP/23exten = 23 State:Unavailable
Watchers 0
22 : SIP/22exten = 22 State:Unavailable
Watchers 0
21 : SIP/21exten = 21 State:Unavailable
Watchers 0
- 4 hints registered
asterisk-er*CLI
The phones register with only one SIP-extension. We use SIP Version
1.5.3 for the phones.
Part from my extension.conf:
exten = 21,hint,SIP/21
exten = 21,1,Macro(stdexten,21,SIP/21,${CALLERIDNUM},${CIDNAME})
Thanks for any help!
Regards,
BK
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[Asterisk-Users] Need help with script from http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
Hi, can anybody help me with the allcall.agi script from http://www.voip-info.org/wiki/view/Polycom+auto-answer+config When I run the script offline from asterisk it seems to work but inside my dialplan it does nothing - it does not write any calling file. -- Executing AGI(SIP/31-ee46, allcall.agi|SIP/31 ) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/allcall.agi -- AGI Script allcall.agi completed, returning 0 Has anybody experience with this? Thanks, BK ___ --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] Octo Bri card together te405p and bristuff
Hi, is it possible to run an octoBri card together with a TE405P card in one system with bristuff? If yes, how should the zaptel.conf look like? Thanks and regards, BK ___ --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] zaptel 1.2.0 and correct settings in zapata.conf for Germany
Hi, everything works fine with zaptel 1.2.0 and TE405P. The only thing i am missing is the callerid for incoming calls. It is always empty. That worked with 1.0.9. -- Accepting overlap call from '' to '9671987' on channel 0/2, span 1 Are there any missing setting in the zapata.conf to make the incoming callerid number visible? Thanks and regards BK ___ --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_misdn crashes : init_stack: success but entitylist not empty
yes, i can confirm this. We had similar problems. FC4 comes with gcc4. We added gcc3 and recompiled the kernel, asterisk and chan_misdn. Now we can load chan_misdn.so with crashing the asterisk server. Johann Steinwendtner wrote: Make sure that you compile misdnuser with gcc3.x, gcc4 did not work for me. Hans Yoann Le Bihan schrieb: Jose, I met so many problems these last 8 days that I don't remember exactly which config was mine at that time, so I can't testify the answer... (just for fun : my linux box is having 3 hd with a different distro on each of them and I plug the cable on the hd I want to boot depending on my mood ;o)). I think I was running 1.0.9. The main things I did were : - deinstalling everything (asterisk, misdn, misdnuser, chan_misdn, ...) - compiling and installing asterisk 1.2.0 (make ; make install) - downloading the install_misdn script on beronet (http://www.beronet.com/download/install-misdn.tar.gz) and executing the make install (be careful : you need kernel headers) And now, I'm done : Asterisk runs without chan_misdn, but crashes with it :-( But it's installed :-) Good luck ! ;) Cheers, YLB. 2005/11/25, Jose Limeres [EMAIL PROTECTED]: Yoann, I am going through a similar problem you reported in a past posting: Nov 24 17:49:31 ERROR[9326] chan_misdn.c: Unable to initialize mISDN Nov 24 17:49:31 WARNING[9326] loader.c: chan_misdn.so: load_module failed, returning -1 Nov 24 17:49:31 WARNING[9326] chan_misdn.c: cb_log called with out-of-range port number! (0) Nov 24 17:49:31 WARNING[9326] loader.c: Loading module chan_misdn.so failed! How did you solve it? Thanks, jose On 25/11/05, Yoann Le Bihan [EMAIL PROTECTED] wrote: Hi, Asterisk 1.2 on FC4, all is right, I'm happy. But when I try to load chan_misdn after a successful install, I get it : # asterisk -vvvgc [...] [chan_features.so] = (Feature Proxy Channel) == Registered channel type 'Feature' (Feature Proxy Channel Driver) [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found Got: 1 from get_ports Init. Stack on port:1 No Connect port:1 init_stack: Success # Nothing else. Asterisk crashes. If I look at /var/log/messages : # tail /var/log/messages Nov 25 00:22:39 toto kernel: Debug: sleeping function called from invalid context at arch/i386/lib/usercopy.c:634 Nov 25 00:22:39 toto kernel: in_atomic():0, irqs_disabled():1 Nov 25 00:22:39 toto kernel: [c01d16b8] copy_from_user+0x18/0x80 Nov 25 00:22:39 toto kernel: [e0ba85b8] mISDN_write+0x318/0x7c5 [mISDN_core] Nov 25 00:22:39 toto kernel: [e0ba82a0] mISDN_write+0x0/0x7c5 [mISDN_core] Nov 25 00:22:39 toto kernel: [c0158bb1] vfs_write+0xa2/0x15a Nov 25 00:22:39 toto kernel: [c0158d14] sys_write+0x41/0x6a Nov 25 00:22:39 toto kernel: [c0102ec5] syscall_call+0x7/0xb Nov 25 00:22:39 toto kernel: MISDN free_device: entitylist not empty # Any idea ?... I've been on it for 1 whole week... I'm exhausted :-( Cheers, YLB. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] How to initiate a call from a web page?
Hi, we have a html based telephonelist on our intranet site. Does there exist any solution to initiate a call from a link ? We use Polycom SIP IP phones. thanks and regards, bk ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Voicemail - new feature request
Hi, I don't if was yet an issue. It really would be nice if each user is able to active/deactivate the mail forwarding of his voicemail via the VoiceMailMenu. Regard ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 Fritz! + chan_capi + mISDN + PTP
I think you can't use a Fritz Card for PTP. You need an active card. We use the
the beronet ISDN Cards with misdn.
Lionel Riem wrote:
Hello everyone,
I have been using an AVM Fritz! card with chan_capi and mISDN for quite
a while in PTM mode and it was working finely.
Now, I needed more DID/MSN, so I switched to PTP. But now nothing works
anymore :(
I am using Asterisk on Debian Sarge stable and installed Asterisk along
with chan_capi from apt-get. I installed mISDN from the CVS of
isdn4linux.de.
It is :
- Asterisk 1.0.7 with bristuff
- chan_capi 0.3.5
When I load the whole modules lot, I get the following in dmesg:
Modular ISDN Stack core $Revision: 1.25 $
mISDNd: kernel daemon started
ISAC module $Revision: 1.16 $
mISDNd: test event done
CAPI Subsystem Rev 1.1.2.8
capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs)
ISDN L1 driver version 1.11
ISDN L2 driver version 1.20
mISDN: DSS1 Rev. 1.30
mISDN Capi 2.0 driver file version 1.14
X25 DTE modul version 1.8
AVM Fritz PCI/PnP driver Rev. 1.30
ACPI: PCI interrupt :00:14.0[A] - GSI 10 (level, low) - IRQ 10
mISDN_fcpcipnp: found adapter Fritz!Card PCI v2 at :00:14.0
fritz card cd09a000 dch cd09a094 bch1 cd09a214 bch2 cd09a3a0
AVM PCI V2: stat 0x240020e
AVM PCI V2: Class E Rev 2
AVM PnP: HDLC version 2
mISDN: AVM Fritz!PCIv2 config irq:10 base:0xEC00
spin_lock_adr=cd09a024 now(d015b867)
busy_lock_adr=cd09a024 now(d015b867)
AVM PCI/PnP: reset
AVM PCI/PnP: S0/S1 40/2
Fritz1 ISAC STAR 40
Fritz1 ISAC MODE c0
Fritz1 ISAC ADF2 ff
Fritz1 ISAC ISTA 0
Fritz1 ISAC CIR0 7
mISDN_isac_init: ISACSX
Fritz1 HDLC 1 STA 8200
Fritz1 HDLC 2 STA 8200
AVM Fritz!PCI: IRQ 10 count 4
fritz 1 cards installed
Here is my /etc/asterisk/capi.conf:
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
mode=immediate
isdnmode=ptp
msn=*
incomingmsn=*
controller=1
softdtmf=1
context=dispatcher
accountcode=
devices=2
Here is my /etc/modprobe.d/capi conf file:
alias /dev/capi20 avmfritz
alias char-major-68-0 avmfritz
install avmfritz /sbin/modprobe capi; \
/sbin/modprobe mISDN_core; \
/sbin/modprobe mISDN_l1; \
/sbin/modprobe mISDN_l2; \
/sbin/modprobe l3udss1; \
/sbin/modprobe mISDN_capi; \
/sbin/modprobe mISDN_x25dte; \
/sbin/modprobe --ignore-install avmfritz protocol=0x22
remove avmfritz /sbin/modprobe -r --ignore-remove avmfritz; \
/sbin/modprobe -r mISDN_x25dte; \
/sbin/modprobe -r mISDN_capi; \
/sbin/modprobe -r l3udss1; \
/sbin/modprobe -r mISDN_l2; \
/sbin/modprobe -r mISDN_l1; \
/sbin/modprobe -r mISDN_core; \
/sbin/modprobe -r capi
capiinfo shows me:
asterisk:/etc/asterisk# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: mISDN CAPI controller Fritz1
CAPI Version: 2.0
Manufacturer Version: 1.0
Serial Number: 0002
BChannels: 2
Global Options: 0x0018
DTMF supported
Supplementary Services supported
B1 protocols support: 0x0003
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
B2 protocols support: 0x0043
ISO 7776 (X.75 SLP)
Transparent
Transparent (ignoring framing errors of B1 protocol)
B3 protocols support: 0x0005
Transparent
ISO 8208 (X.25 DTE-DTE)
0100
0200
1800
0300
4300
0500
Supplementary services support: 0x0012
Terminal Portability
Call Forwarding
In Asterisk, when an incoming call arrives, it shows me the following:
Asterisk Ready.
*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
*CLI capi debug
CAPI Debugging Enabled
*CLI
*CLI
*CLI -- INFO_IND ID=001 #0x0001 LEN=0016
Controller/PLCI/NCCI= 0x101
InfoNumber = 0x18
InfoElement = 89
-- INFO_IND ID=001 #0x0001 LEN=0016
Controller/PLCI/NCCI= 0x101
InfoNumber = 0x18
InfoElement = 89
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1184 find_pipe: PLCI doesnt
match last pipe (PLCI = 0x101)
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1301 pipe_msg: INFO_IND
ID=001 #0x0001 LEN=0016
Controller/PLCI/NCCI= 0x101
InfoNumber = 0x18
InfoElement = 89
-- CONNECT_IND ID=001 #0x0002 LEN=0044
Controller/PLCI/NCCI= 0x101
CIPValue= 0x1
CalledPartyNumber = 8120
CallingPartyNumber = 01 830123456789
CalledPartySubaddress = default
CallingPartySubaddress = default
BC = 80 90 a3
LLC = default
HLC = default
AdditionalInfo
BChannelinformation= default
Keypadfacility = default
Useruserdata = default
Facilitydataarray = default
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1931
[Asterisk-Users] Where to get the latest SIP Firmware for Polycom Phones?
Hello, can anybody tell me where to get the latetest SIP Firmware 1.6.2 for the Polycom phones? Thanks and regards, KB ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SIP Attended Transfer using REFER and Replaces: headers
Hi, sorry, but does this belong to the issue Attended transfer with original CID info? Will this work with a future release? Thanks kb Olle E. Johansson wrote: Dinesh Nair wrote: hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. That is not supported today. However, I have working code that will be submitted to the bug tracker after Astricon. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. is this the way it's supposed to work ? Yes. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SIP Attended Transfer using REFER and Replaces: headers
Is there a schedules for this? Olle E. Johansson wrote: This work will belong to a future version of Asterisk, not 1.2 release. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Recommendation for 8 lines analog card in Australia
Hi, we want to build a Asterisk server for a branch office in Australia. At the moment they use 5 analog lines. We will need at least 8 lines. What hardware would you recommend for the 8 analog PSTN lines? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] TE405P V2 changes?
Hi, we also got one V2 TE405P card. It works fine now. At the moment we use for bridging the Pri to our old PBX. You must use zaptel, libri and asterisk v 1.0.8 or higher. We use 1.0.9 at the moment. zaptel: after make; make install i also executed make config. This copies the correct startup script to /etc/init.d/zaptel. Without this it also didn't worked for me. Master Abi wrote: Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with 2.6.12 kernel Zapata Telephony Interface Registered on major 196 Found TE4XXP at base address fdfff000, remapped to f8928000 TE4XXP version c01a0164, burst ON, slip debug: OFF TE4XXP running with work queues. FALC version: 0005, Board ID: 00 Reg 0: 0x364e9400 Reg 1: 0x364e9000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE405P (2nd Gen) eth0: link up, 10Mbps, half-duplex, lpa 0x About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Unassigning channel 0/1! Unassigning channel 0/2! Unassigning channel 0/3! Unassigning channel 0/4! Unassigning channel 0/5! Unassigning channel 0/6! Unassigning channel 0/7! Unassigning channel 0/8! Unassigning channel 0/9! Unassigning channel 0/10! Unassigning channel 0/11! Unassigning channel 0/12! Unassigning channel 0/13! Unassigning channel 0/14! Unassigning channel 0/15! Unassigning channel 0/16! Unassigning channel 0/17! Unassigning channel 0/18! etc... This was working for 10 months before the upgrade. Master ___ 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] TE405P V2 changes?
yes, fedora 3 but without any changes at the sources Master Abi wrote: Are you using Redhat/Fedora? If I remember those init scripts is for Redhat/Fedora. I am using gentoo. Did you make any modifications to wct4xxp.c. or pass any parameters to zaptel. I see there is a #define SUPPORT_GEN1 in to wct4xxp.c which I commented out, but it made no difference. ztcfg seems to where the channels become unassigned. Thanks again. Kib Eki wrote: Hi, we also got one V2 TE405P card. It works fine now. At the moment we use for bridging the Pri to our old PBX. You must use zaptel, libri and asterisk v 1.0.8 or higher. We use 1.0.9 at the moment. zaptel: after make; make install i also executed make config. This copies the correct startup script to /etc/init.d/zaptel. Without this it also didn't worked for me. Master Abi wrote: Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with 2.6.12 kernel Zapata Telephony Interface Registered on major 196 Found TE4XXP at base address fdfff000, remapped to f8928000 TE4XXP version c01a0164, burst ON, slip debug: OFF TE4XXP running with work queues. FALC version: 0005, Board ID: 00 Reg 0: 0x364e9400 Reg 1: 0x364e9000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a0164 Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE405P (2nd Gen) eth0: link up, 10Mbps, half-duplex, lpa 0x About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Unassigning channel 0/1! Unassigning channel 0/2! Unassigning channel 0/3! Unassigning channel 0/4! Unassigning channel 0/5! Unassigning channel 0/6! Unassigning channel 0/7! Unassigning channel 0/8! Unassigning channel 0/9! Unassigning channel 0/10! Unassigning channel 0/11! Unassigning channel 0/12! Unassigning channel 0/13! Unassigning channel 0/14! Unassigning channel 0/15! Unassigning channel 0/16! Unassigning channel 0/17! Unassigning channel 0/18! etc... This was working for 10 months before the upgrade. Master ___ 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 ___ 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
[Asterisk-Users] Problem with setting the right dialplan for german PRI E1 on TE405P from digium
Hi, I tried so many but can't find the right setting for my problem. What do i have to configure so that the complete number including extension is displayed at the called party. At the moment the called party only sees the number 7837-0 not the 7837-134. Everything works fine. Incoming and Outgoing calls. Is there someone who configured a german pri with that digium card? I really appreciate any help. Thanks ___ 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] Digium TE405P, caller id and migration to *
Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 - * routes to old pbx - old pbx sees 123456 as caller number. 2. A call made from a SIP client to the outside lacks the extension in the number: Ex: PSTN number is 6789-0. The extension 234 is not added to the PSTN number like 6789-234 when dialing out over the PSTN. Can anybody tell me how i must change the configuration? Do you need the zapata.conf? Thanks in advance and regards ___ 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] Digium TE405P, caller id and migration to *
Andrew Kohlsmith wrote:
On Monday 08 August 2005 04:03, Kib Eki wrote:
1. A call from the outside to the old PBX is missing a leading 0 before the
number. Ex: caller has number 0123456 - * routes to old pbx - old pbx
sees 123456 as caller number.
This is absolutely trivial to fix. Anyone who's been able to put * between a
PRI and a PBX should be able to figure this out without asking the list.
It's trivial dialplan stuff.
exten = _X.,1,Dial(Zap/g2/0${EXTEN}) kind of trivial. You may have to debug
a little to see where or why the 0's disappearing.
Misunderstanding: I need to change the calleridnum because there is missing the
0 before the number.
2. A call made from a SIP client to the outside lacks the extension in the
number: Ex: PSTN number is 6789-0. The extension 234 is not added to the
PSTN number like 6789-234 when dialing out over the PSTN.
Again, trivial dialplan stuff. Your sip.conf will have the callerid for each
SIP client and you can append that information to the outgoing CID.
That is set correctly and works between sip clients. it is only a problem when i
try to dial out over zap/g1.
-A.
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Re: [Asterisk-Users] Digium TE405P, caller id and migration to *
Peter Svensson wrote: On Mon, 8 Aug 2005, Kib Eki wrote: Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 - * routes to old pbx - old pbx sees 123456 as caller number. See internationalprefix, nationalprefix etc in the file zapata.conf. 2. A call made from a SIP client to the outside lacks the extension in the number: Ex: PSTN number is 6789-0. The extension 234 is not added to the PSTN number like 6789-234 when dialing out over the PSTN. Are you refering to the dialed number or the outgoing caller id (calling number)? yes i refering to the my outgoing number (caller id) Peter ___ 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] Digium TE405P, caller id and migration to *
Andrew Kohlsmith wrote:
On Monday 08 August 2005 10:56, Kib Eki wrote:
Misunderstanding: I need to change the calleridnum because there is missing
the 0 before the number.
SetCIDNum(0${CALLERIDNUM}) or something?
yes, but that does not work the zap channel connected the pbx. means i had no
success with this
That is set correctly and works between sip clients. it is only a problem
when i try to dial out over zap/g1.
Are you mangling the outoging caller ID in your Zap-terminating extension
contexts?
Yes.
-A.
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Re: [Asterisk-Users] Music on Hold: CPU Intensive Monster
Hi Matthew, i found the following link very usefull: http://www.orderlyq.com/asteriskqueues.html#moh It is an alternativ to mpg123. It works very fine for me. Regards Matthew Boehm wrote: OK. So I did a test last night. All of asterisk's threads where using 0.0% CPU. I made 1 call to our call queue. CPU jumped to average of 9% and stayed around that for the 2 minutes I was in the queue just listening to music on hold. MOH is in MP3 format and I'm using format_mp3. Phone was linksys PAP2-NA using G729. Can I reasonably assume that the 9% was decoding the MP3, then encoding G729? I tried using Anthm's RAW format but that actually made things worse. I tried going back to mpeg321 and asterisk still used the same amount of CPU. Any ideas for getting processor usage down on MOH? -Matthew ___ 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
[Asterisk-Users] New feature in V1.2: attended call transfer
Olle, thank you very much for the summary of the new features in 1.2. Concerning the new feature attended call transfer: Is it implemented that the original caller id will be passed to transferee? like this: A calls BB transfers to CC see a call from A. Regards ___ 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] cdr_mysql does not write to mysql db
Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards ___ 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] cdr_mysql does not write to mysql db - SOLVED
i reinstalled the addons and the module works fine now. Thanks to all!! Neal Lawson wrote: using localhost in you mysql conf should work, make sure you linux box and the loopback interface up and has a a entry in your /etc/ hosts for localhost and that your firewall (if you have one setup on your localbox) allows traffic from 172.0.0.1 to 172.0.0.1 On Jul 27, 2005, at 6:17 AM, Dpto. Técnico. wrote: Try to put the IP of you CDR server instead of 'localhost', that's work for me. Regards. - Original Message - From: Kib Eki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 27, 2005 9:44 AM Subject: [Asterisk-Users] cdr_mysql does not write to mysql db Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards ___ 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 ___ 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
[Asterisk-Users] Meetme and option c for announcing user count
Hi, the option c for the announce of the user count does not work me in * 1.0.9. exten = ,1,Wait(1) exten = ,2,MeetMe(|Mdcs) And how to handel the marked mode with option A? I can't find any sample config for this. Regards ___ 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] Asterisk Configuration
Hi Afzall, i am also still a beginner on *. A made best experience with the * wiki on http://www.voip-info.org/wiki-Asterisk. Maybe you start with the introduction part. Afzaal Mirza wrote: Dear users, I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. Regards, Afzaal ___ 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
[Asterisk-Users] Which ATA adapter to use with an analog fax maschine?
Hi, i need an recommandation for an ATA adapter to use with an anlog fax maschine. I would appreciate any hints. Regards! ___ 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] Attended transfer with original CID info?
Hi, is it possible to do an attended transfer so that the original CID info will stay for that call. With blind transfer this works. Regards ___ 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] Meet Me - this is not a valid conference number, please try again
will the lsmod list show you ztdummy modul? if not, modprobe ztdummy I think without a timer source meetme won't work Erdem HAKİ wrote: Hello, I’m trying Meet Me Feature. I read wiki , searched google and i configured my extension.conf and meetme.conf. But I receive “this is not a valid conference number, please try again” message, so what could be the problem? Thanks for your interest. Erdem HAKI ___ 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
[Asterisk-Users] How to integrate the Call Pickup with CID info feature in the release tree of Asterisk?
Hi, we really need the feature Call Pickup with CID info http://www.voip-info.org/wiki-Asterisk+bounty+Call+Pickup+with+CID+info.+-+SIP in the current Asterisk release because we have a newer TE405P card which needs 1.0.8 or newer to work. The call pickup patch only works for 1.0.7. Who is responsible for such a wish? Regards, Kib ___ 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] No channels after starting asterisk
Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: Individual Clear channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: Individual Clear channel (Default) (Slaves: 07) Channel 08: Individual Clear channel (Default) (Slaves: 08) Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: Individual Clear channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: Individual Clear channel (Default) (Slaves: 14) Channel 15: Individual Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) Channel 18: Individual Clear channel (Default) (Slaves: 18) Channel 19: Individual Clear channel (Default) (Slaves: 19) Channel 20: Individual Clear channel (Default) (Slaves: 20) Channel 21: Individual Clear channel (Default) (Slaves: 21) Channel 22: Individual Clear channel (Default) (Slaves: 22) Channel 23: Individual Clear channel (Default) (Slaves: 23) Channel 24: Individual Clear channel (Default) (Slaves: 24) Channel 25: Individual Clear channel (Default) (Slaves: 25) Channel 26: Individual Clear channel (Default) (Slaves: 26) Channel 27: Individual Clear channel (Default) (Slaves: 27) Channel 28: Individual Clear channel (Default) (Slaves: 28) Channel 29: Individual Clear channel (Default) (Slaves: 29) Channel 30: Individual Clear channel (Default) (Slaves: 30) Channel 31: Individual Clear channel (Default) (Slaves: 31) Channel 32: Individual Clear channel (Default) (Slaves: 32) Channel 33: Individual Clear channel (Default) (Slaves: 33) Channel 34: Individual Clear channel (Default) (Slaves: 34) Channel 35: Individual Clear channel (Default) (Slaves: 35) Channel 36: Individual Clear channel (Default) (Slaves: 36) Channel 37: Individual Clear channel (Default) (Slaves: 37) Channel 38: Individual Clear channel (Default) (Slaves: 38) Channel 39: Individual Clear channel (Default) (Slaves: 39) Channel 40: Individual Clear channel (Default) (Slaves: 40) an so on for rest of the channels Tom Hayden wrote: What kind of output do you get with ztcfg -vv ?? -- Tom On 7/13/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Jul 13, 2005 at 05:19:08PM +0200, Kib Eki wrote: Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [pstn] Shouldn't that be [channels] ? Why can't i see or use my channels? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] No channels after starting asterisk
Bob, there are no error messages. This is the first time we installed a TE405P adapter to the system. So that is the change to the system. Bob Goddard wrote: On Wednesday 13 Jul 2005 16:19, Kib Eki wrote: Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [...] Why can't i see or use my channels? You're not going to get anywhere unless you show us the error messages and what if anything has changed on your system. B ___ 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] No channels after starting asterisk - SOLVED!!!
That what was exactly the mistake in the configuration. I changed [pstn] to [channels] and restartet *. Thank you very much! Tzafrir Cohen wrote: On Wed, Jul 13, 2005 at 05:19:08PM +0200, Kib Eki wrote: Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [pstn] Shouldn't that be [channels] ? Why can't i see or use my channels? ___ 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] No more sound on MOH after adding TE405P
Hi, after we successfully installed the TE405P card (thanks to this list) the musiconhold does not work anymore. Asterisk starts the mpg123 programm but there is no sound we can hear. Thanks, Kib ___ 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] No channels after starting asterisk
Hi, i am running * 1.0.9 with a newer version of the TE405P. Modprobe wct4xxp and ztcfg are OK. zap show channels only shows me the following. my zapata.conf: [pstn] switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley could be produce hangups callwaitingcallerid=yes ; show callerid on callwaitingcalls echotraining=no echocancel=no echocancelwhenbridged=no overlapdial=yes immediate=no callerid=asreceived language=de rxgain=0.0 txgain=0.0 group=1 signalling=pri_cpe context=incoming channel = 1-15,17-31 group=2 signalling=pri_net context=outgoing channel =32-46,48-62 Why can't i see or use my channels? thanks, Kib ___ 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] Meet Me Configuration
do you have a zap module installed? if not you must run ztdummy as a timer interface Cavanna, Richard wrote: I am trying to configure MeetMe so that external callers can enter the conference rooms after an IVR menu. I have created Conf rooms for all internal Ext’s with a prefix of 8. When I call into the system from my vonage trunck the IVR picks up but will not let me dial a conf room. It tells me it is a invalid extension. Can anyone help with a sample conf on this? Thanks, RC ___ 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
[Asterisk-Users] How to integrate the Call Pickup with CID info feature in the release tree of Asterisk?
Hi, we really need the feature Call Pickup with CID info http://www.voip-info.org/wiki-Asterisk+bounty+Call+Pickup+with+CID+info.+-+SIP in the current Asterisk release because we have a newer TE405P card which needs 1.0.8 or newer to work. The call pickup patch only works for 1.0.7. Who is responsible for such a wish? Regards, Kib ___ 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] No sound when dialing out over SIP Proxy
Hi,
i have trouble to dial out over my sip-provider gmx.
I can register with my provider over port 5060 and also dial out.
It rings at the remote phone but when the call is answered there is no
sound / voice to hear.
This is the part from my sip.conf and extensions.conf:
register = 12345:[EMAIL PROTECTED]
[gmx-out]
type=peer
secret=12345
username=12345
host=sip.gmx.net
fromuser=12345
fromdomain=sip.gmx.net
disallow=all
allow=alaw
allow=ulaw
allow=g729
exten = _0.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Thanks,
kib
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[Asterisk-Users] How to fetch a call not in the same callgroup
Hi, the situation: A call rings at extension 123. My own extension is not in the same call- or pickupgroup for that extension. Is there a way to route the ringing extension 123 to my phone? Thanks, Kib ___ 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] Help with Dial multiple channels simultanously
Hi, the following from extension.conf does not work correctly: exten = 301, 1, Dial(SIP/455SIP/456, 15) That is the console output: -- Executing Dial(mISDN/1/105, SIP/455SIP/456SIP/456| 10) in new stack -- Called 455 -- Called 456 -- SIP/455-46a8 is ringing == Spawn extension (incoming, 301, 1) exited non-zero on 'mISDN/1/105' As you can see only the extension 455 is dialed. What is wrong with my configuration? Thank you very much, Kib ___ 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] Polycom display variable
Hi, does anyone know what Asterisk variable must be set to manipulate the line under From:-line with a polycom 500 ip phone? Thanks + regards, Kib ___ 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] Help with Dial multiple channels simultanously
yes, you are right - the extension.conf wasn't the same as debug output but it is solved anyway. There was just a missing registration for the extension 456 Thanks Asterisk wrote: Something is not quite right - your extensions.conf is specifying Dial(SIP/455SIP/456, 15) but the console is showing Executing Dial(mISDN/1/105, SIP/455SIP/456SIP/456| 10) note the extra SIP/456 (as in SIP/456SIP/456) and the 10 instead of the 15 in the extensions.conf. Are you sure you've posted the correct extensions.conf ? Julian Kib Eki wrote: Hi, the following from extension.conf does not work correctly: exten = 301, 1, Dial(SIP/455SIP/456, 15) That is the console output: -- Executing Dial(mISDN/1/105, SIP/455SIP/456SIP/456| 10) in new stack -- Called 455 -- Called 456 -- SIP/455-46a8 is ringing == Spawn extension (incoming, 301, 1) exited non-zero on 'mISDN/1/105' As you can see only the extension 455 is dialed. What is wrong with my configuration? Thank you very much, Kib ___ 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 ___ 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] Polycom display variable
This works for me:
to display the following on the polycom phone:
From: Support-Group
x
--- the caller id number
you can use the following code in extension.conf:
exten = 301, 1, Dial(SIP/456SIP/455SIP/457, 30)
exten = 301, 2, SetVar(foo=* Support-Group * ${CALLERIDNUM})
exten = 301, 3, SetCallerID(${foo})
exten = 301, 4, Dial(SIP/705)
exten = 301, 5, Hangup
Kib Eki wrote:
Hi,
does anyone know what Asterisk variable must be set to manipulate the
line under From:-line with a polycom 500 ip phone?
Thanks + regards,
Kib
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[Asterisk-Users] misdn and call hangup problem
Hi,
we test the misdn module together with beronet BN8S0 card.
We connect the pstn ISDN line to Port 1 and an ISDN phone to Port 2.
That works great, the ISDN phone rings an we can make the call.
When the caller hangsup before call is answered by the callee the call
on Port 2 rings until end of day.
This is the extensions.conf part for this:
[incoming]
exten = _., 1, Dial(mISDN/g:ntports/${EXTEN})
exten = _., 2, Congestion
[outgoing]
exten = _., 1, Dial(mISDN/g:teports/${EXTEN})
exten = _., 2, Congestion
This problem does not occur when we call the isdn phone from a sip client.
Can anybody tell what is wrong with this configuration.
Thanks,
Kib
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[Asterisk-Users] No mans problem?
Hi, i try again to ask this. When i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first only rings at my phone and in the second i can see the callers number before i am connected. I am using a polycom 500 ip phone. Is this a special polycom problem? Thanks, Kib ___ 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] Need Help with pickup *8
Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first only rings at my phone and in the second i can see the callers number before i am connected. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib ___ 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] Pickup problem
Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first rings at my phone and in the second i can see the callers number. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib ___ 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] Digium G729 licensing - is it worth the trouble?
We want to build the new Asterisk PBX with the Polycom 500 IP phone. So G.729 is the only alternative for small codec for WAN calls, isn't it? Brian McSpadden wrote: On 6/5/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote: I think you are at least morally correct, I think I might do that. However, I guess part fo the question is, does it make much difference, am I that badly off without G729? The other point it, doesn't Digium realize they are pissing customers off with their attitude? I took a lot of my time to explain my situation, send them a letter, call several times, and they still won't allow me to use what I paid for. Is it my fault they have a stupid and unworkable enforcement system? Is it more important to prevent piracy or keep customers? I think they have it backwards. They're also really bad about supporting the activation of the codecs when you do buy them...I bought 6 of them (3 for each site in this case), and tried to activate them through the customer's very restrictive firewall. Of course it didn't work since the firewall doesn't allow arbitrary port numbers to leave the network, and they have never even returned my calls on the issue. It wouldn't really bother me as much if they'd at least have had the courtesy to call me and tell me it was not possible to activate them over some other method (manual, http, https, etc). I've still got all 6 of them completely unused because of that. I'm now very selective on when and if I'll use g.729. My advice is to use anything else first, and use g.729 as a last resort. Brian ___ 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] Digium G729 licensing - is it worth the trouble?
could you please explain the transcoding to me? How do i have to configure this? Chris Mason (Lists) wrote: Yes, I use the phones on a LAN and don't care about the bandwidth, then allow Asterisk to transcode to GSM for trunked calls. I was using G729 all the way, but the licensing stuff caused me too many problems. WHen I messed up the licensing by allowing the order of the modules to be reversed, thereby putting eth0 and eth1 on different NICs, it was a holiday weekend, and I was not able to get anyone at Digium until Tuesday. They still would not permit the relicensing. We ended up three days without the codecs we paid for, and so I had to re-engineer the system, moving the phones to uLaw and usinggsm at both ends of the trunk. For reliability reasons I would not advise g729. If you lose a NIC, or you are using the motherboard's interface and it fails, you will have to relicense. If that fails, you have to restructure everything. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kib Eki Sent: Monday, June 06, 2005 3:22 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Digium G729 licensing - is it worth the trouble? We want to build the new Asterisk PBX with the Polycom 500 IP phone. So G.729 is the only alternative for small codec for WAN calls, isn't it? Brian McSpadden wrote: On 6/5/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote: I think you are at least morally correct, I think I might do that. However, I guess part fo the question is, does it make much difference, am I that badly off without G729? The other point it, doesn't Digium realize they are pissing customers off with their attitude? I took a lot of my time to explain my situation, send them a letter, call several times, and they still won't allow me to use what I paid for. Is it my fault they have a stupid and unworkable enforcement system? Is it more important to prevent piracy or keep customers? I think they have it backwards. They're also really bad about supporting the activation of the codecs when you do buy them...I bought 6 of them (3 for each site in this case), and tried to activate them through the customer's very restrictive firewall. Of course it didn't work since the firewall doesn't allow arbitrary port numbers to leave the network, and they have never even returned my calls on the issue. It wouldn't really bother me as much if they'd at least have had the courtesy to call me and tell me it was not possible to activate them over some other method (manual, http, https, etc). I've still got all 6 of them completely unused because of that. I'm now very selective on when and if I'll use g.729. My advice is to use anything else first, and use g.729 as a last resort. Brian ___ 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 ___ 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] Automatic Codec change for different communication channels!?
Hi, I am looking for a way to let * choose the voice codec relying to the used communication channel. Example I am using a Polycom 500 which supports G729 and G.711. When I am doing internal calls (with my LAN) or calls over the PSTN (ISDN) I want to use the G.711 codec because there is enough bandwith. When I am doing inter asterisk calls (over my WAN to another * server) I want to use G.729. Is there a way how i can achieve this? Kib ___ 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] Automatic Codec change for different communication channels!?
could you please give more information concerning this setting? Pavel Jezek wrote: you can try use variable preffered_codec in dial command (if you now the prefixes/dial numbers, for which to use eg. g729)... PJ Kib Eki wrote: Hi, I am looking for a way to let * choose the voice codec relying to the used communication channel. Example I am using a Polycom 500 which supports G729 and G.711. When I am doing internal calls (with my LAN) or calls over the PSTN (ISDN) I want to use the G.711 codec because there is enough bandwith. When I am doing inter asterisk calls (over my WAN to another * server) I want to use G.729. Is there a way how i can achieve this? Kib ___ 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 ___ 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] VoiceMail with Polycom 500
Hi, I want to use the sip extension 105 as the voicemailbox number. When i initiate a call to the number 105 from my polycom 105 i only get a call on new line to phone. But i want in this moment is the voicemailmenu which ask me for my password. How can this be done with the polycom phone? Kib ___ 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] snom mass deployment (probably off topic)
David,
this is my config via DHCP:
67: Startupserverwebserver.mydomain.com
66: Startup filesnomstartup.cfg
File snomstartup.cfg
setting_server:
http://webserver.mydomain.com/snom/conf/snomcfg.php?MAC={mac}
subscribe_config: on
File snomcfg.php
?php
$filename = '/var/www/html/snom/conf/snom360.php';
readfile(htmlspecialchars($filename));
$filename = '/var/www/html/snom/conf/snom-'.$MAC.'.cfg';
readfile(htmlspecialchars($filename));
?
Kib
David John Walsh wrote:
Hello
Although not stictly a asterisk issue, any help would be apreciated.
Firstly a few notes on the snom 360, which I have had on a test bed
for the last week. Its a great phone, with a good user interface,
both physically and its web based one.
At its lastest firmware it does have a few quirks, with regards to the
way it handles usernames and passwords on the physical interface.
These have been passed back, and hopefully will be addressed.
Its worst feature as I see it is twofold, with regards to its power
fail features. If it loses power for more than a few minuites it
loses its settings - not the best thing in a world where routers and
firewalls can be given power back days later and be fine.
It has an interesting configuration mode, it tries to contact snom,
who then (if told about it) goes to their national distrubtor who then
either has your config or passes it on again
The settings file is well documented, and you can pull them direct
from phone in a ready to go way.
---
I now have my configs in the file name format of snom360-{mac}.htm
(where {mac} is the MAC address of the phone in question)
The phone initally tries to goto
provisioning.snom.com/snom360/snom360.html this sends it onto
http://snom.com/snom360/snom360.php?mac={mac}
Assuming that I perform some creative dns records on my dns server,
would someone be kind enough to write some sample php code to take the
url
http://snom.com/snom360/snom360.php?mac={mac}
and provide the url http://asterisk-demo/snom/snom360-{mac}.html
The code the url needs to go in is as follows:
# Redirect all phones to the php script
setting_server: http://asterisk-demo/snom/snom360-{mac}.html
I'm useless with php and most launguages, so thank you to any help
this request generates
David
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[Asterisk-Users] Looking for list with asterisk default extensions
Hi, some days ago i found a list with default extensions for things like 'echo test = *44' . But i can't the point where they have been. So, maybe someone of knows what i looking for. thanks, kib ___ 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] Transparently Routing German pri through Asterisk
Hi, at the moment we have in Avaya Integral PBX with german pri (30 lines). We want to smouthly migrate to an Asterisk server. For this reason: Is it possible to route the external german pri (E1) through Asterisk server to that Avaya PBX? I think at first we need a Digium e1 card 4-Port. But how do we have to configure the routing of the whole PRI? I really would appreciate any sample config. Thanks, Kib ___ 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] Difference between Asterisk and Asterisk@home?
Hi, can one summarize the main differences between Asterisk and [EMAIL PROTECTED] or point me to a location where i can find such a list? Much thanks, Kib ___ 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] X-Lite and callto:// syntax in webpages
Hmmm... Firefox tries to open an external application but nothing happens for IE the protocoll is unknown Any hint? Roman Zhovtulya wrote: I think you should use the sip://name syntax. I've wasted a lot of time before I figured it out myself. Regards, Roman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Kib Eki Sent: Montag, 2. Mai 2005 14:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] X-Lite and callto:// syntax in webpages Hi, does anyone know if x-lite supports the callto://name syntax on web pages as skype does? Kib ___ 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 ___ 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] Is there any chance to bring Skype and Asterisk User together?
Hi, is there any chance to bring Skype and Asterisk User together? Regards, Kib ___ 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] X-Lite and callto:// syntax in webpages
Hi, does anyone know if x-lite supports the callto://name syntax on web pages as skype does? Kib ___ 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] the beginning of voice menu is cutted
Hi, when I dial my voicemenu the menu voice is always cutted so that i only hear 'word from password. What do i have to configure so that i hear the full text from the beginning? thanks, Kib ___ 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] No busy tone when dialing out over ISDN with Polycom 500 IP
Isn't the zapata.conf only for Digium hardware? I use an Eicon Card. Eric Wieling aka ManxPower wrote: Kib Eki wrote: Hi, what do i have to configure to get a busy tone when dialing out over ISDN channel with my Polycom 500 IP? Try priindication = inband in /etc/asterisk/zapata.con ___ 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
[Asterisk-Users] No busy tone when dialing out over ISDN with Polycom 500 IP
Hi, what do i have to configure to get a busy tone when dialing out over ISDN channel with my Polycom 500 IP? Kib ___ 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] Busy line status and chan_capi?
What do i have to confiure so that a call comming in the * server through chan_capi recognizes a normal busy line beep if the SIP phone is busy? Kib ___ 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] Busy line status and chan_capi?
Elmar, I tried the config from Damian and works for me. The only problem is that it is not a traditional german busy tone but an american one. Maybe there is also a solution to this. Kib Elmar Haneke wrote: What do i have to confiure so that a call comming in the * server through chan_capi recognizes a normal busy line beep if the SIP phone is busy? Presumably you have to fix the code and recompile chan_capi. I did try the same without any success, I'm shure that it's an chan_capi bug. The only method I found to prevent the false ringing indication is to use hangup to reject the call. Elmar ___ 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
[Asterisk-Users] Hylafax and Asterisk
Hi, I found the following from the wiki: ** HylaFax and Asterisk Another solution is the Hylafax software. capi4hylafax and chan_capi will gladly coexist. You just tell asterisk to ignore the DIDs that are used for fax. My question: How can I tell * to ignore special DIDs and let them through to Hylafax? Kib ___ 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] Busy line status and chan_capi?
What do i have to confiure so that a call comming in the * server through chan_capi recognizes a normal busy line beep if the SIP phone is busy? Kib ___ 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] Voicemail and SJphone
Hi, what configuration on asterisk is missing when SJphone tells me that the voicemail number is not configured? current config: sip.conf [EMAIL PROTECTED] voicemail.conf 102 = 102,Kib, [EMAIL PROTECTED] What is missing? Regards, Kib ___ 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] Voicemail and SJphone
Thanks, I entered the voicemail number to SJphone clicked the mailbox button. The programm dials but nothing happens. Rikard Westlund wrote: If your voicemail is setup correctly then you need to klick on the options icon on your sjphone and go to profiles. Choose the profile that you are using and click edit. Then go to General and type the extension that you use in the voicemail address field. Cheers Rikard -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kib Eki Sent: den 6 april 2005 09:09 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail and SJphone Hi, what configuration on asterisk is missing when SJphone tells me that the voicemail number is not configured? current config: sip.conf [EMAIL PROTECTED] voicemail.conf 102 = 102,Kib, [EMAIL PROTECTED] What is missing? Regards, Kib ___ 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 ___ 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] Problem with dial out via chan_capi
Hi *,
we successfully integrated the eicon diva 4 bri card in our Asterisk system.
I can dial in to system and route to sip peers.
I tried to dial out with following configuratin without any luck:
extensions.conf:
exten = _5.,1,Dial(CAPI/@301:b${EXTEN})
capi.conf:
[general]
mode=immediate
isdnmode=multipoint
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=301
incomingmsn=*
controller=2
context=default
echocancel=1
echotail=64
devices=2
Console output as follow:
-- Executing Dial(SIP/bdk-d27c, CAPI/@301:b5030225476) in new stack
-- Called @301:b5030225476
-- Setting up echo canceller (PLCI=0x102, function=1, options=2, tail=64)
-- Echo canceller successfully set up (PLCI=0x102)
-- CAPI Hangingup
== No one is available to answer at this time
Can you help me or give me tips?
Thanks in advance.
Kib
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[Asterisk-Users] [Fwd: Problem with dial out via chan_capi]
Hi,
problem solved, I found somethind in this mailing list!
extensions.conf:
exten = 0237482,1,Dial,CAPI/@301:0237482,5,tr
Regards,
Kib
---BeginMessage---
Hi *,
we successfully integrated the eicon diva 4 bri card in our Asterisk system.
I can dial in to system and route to sip peers.
I tried to dial out with following configuratin without any luck:
extensions.conf:
exten = _5.,1,Dial(CAPI/@301:b${EXTEN})
capi.conf:
[general]
mode=immediate
isdnmode=multipoint
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=301
incomingmsn=*
controller=2
context=default
echocancel=1
echotail=64
devices=2
Console output as follow:
-- Executing Dial(SIP/bdk-d27c, CAPI/@301:b5030225476) in new stack
-- Called @301:b5030225476
-- Setting up echo canceller (PLCI=0x102, function=1, options=2, tail=64)
-- Echo canceller successfully set up (PLCI=0x102)
-- CAPI Hangingup
== No one is available to answer at this time
Can you help me or give me tips?
Thanks in advance.
Kib
---End Message---
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[Asterisk-Users] Re: Eicon Diva Server BRI Setup
Hi,
try this:
mknod /dev/capi20 c 68 0
chmod 660 /dev/capi20
I have same configuration as you. It worked for me since yesterday.
Regards,
Kib
---BeginMessage---
Hi *,
we successfully integrated the eicon diva 4 bri card in our Asterisk system.
I can dial in to system and route to sip peers.
I tried to dial out with following configuratin without any luck:
extensions.conf:
exten = _5.,1,Dial(CAPI/@301:b${EXTEN})
capi.conf:
[general]
mode=immediate
isdnmode=multipoint
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=301
incomingmsn=*
controller=2
context=default
echocancel=1
echotail=64
devices=2
Console output as follow:
-- Executing Dial(SIP/bdk-d27c, CAPI/@301:b5030225476) in new stack
-- Called @301:b5030225476
-- Setting up echo canceller (PLCI=0x102, function=1, options=2, tail=64)
-- Echo canceller successfully set up (PLCI=0x102)
-- CAPI Hangingup
== No one is available to answer at this time
Can you help me or give me tips?
Thanks in advance.
Kib
---End Message---
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[Asterisk-Users] Eicon Diva Server BRI Setup
Hi, try this: mknod /dev/capi20 c 68 0 chmod 660 /dev/capi20 I have same configuration as you. It worked for me since yesterday. Regards, Kib ___ 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: Problem with dial out via chan_capi
Thanks, problem solved, I found somethind in this mailing list! Wrong extensions.conf entry. extensions.conf: exten = 0237482,1,Dial,CAPI/@301:0237482,5,tr ?? But, what does ,5,tr mean ?? Regards, Kib ___ 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] Compilation problem chan_capi and Eicon Diva 4Bri
Hi *, I want to integrate the Eicon Diva 4Bri Card to Asterisk. Eicon drivers and capi is installed. I use the latest dev version from eicon compiled and installed for my fedora 2 system. I found the chan_capi for asterisk from www.junghanns.net. Also loaded the patch and applied to the chan_capi source tree. I changed the Makefile to include the capi20.h from eicon: INCLUDE=-I$(ASTERISK_HEADER_DIR) -I/usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon make install gives me the following errors: gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -I/usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c In file included from chan_capi.c:35: /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:52: error: Syntaxfehler before word /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:52: Warnung: kein Semikolon am Ende von »struct« oder »union« /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:53: Warnung: type defaults to `int' in declaration of `maxLogicalConnection' /usr/src/linux-2.6.5-1.358/drivers/isdn/hardware/eicon/capi20.h:53: Warnung: data definition has no type or storage class Unfortunately it is german system so also the compiler errors are in german. I realy need help because I am not the r+d expert. Thanks in advance. Kib ___ 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
