Re: [Asterisk-Users] How to check if a phone / line is used?
In the dialplan you can use ChanIsAvail command Show channels? On Mar 31, 2006, at 2:09 AM, Ronald Wiplinger wrote: In the past I used SetGroup and CheckGroup to figure out if my allowed providers lines are all used or not. Since most of my provider have given me a single line anyway, I wonder if there is a way to check if this (provider) line is taken already. How can I do that? Same is with the phone. How can I see in CLI if a phone is now in use or not? Sip show peers shows me just if it is on-line, but not if it is in a call or not. In the dialplan I could dial the number and if it is busy, it would go to the Voicemail for unavailable or busy. I expect that there is just a test function as well, without trying to call. bye Ronald Wiplinger ___ ___ --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] SIP T
Anyone know how I can get SIP T working w/ Asterisk? TIA, Jon ___ --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] New SkypeSIP gateway
Message: 24 Date: Mon, 03 Apr 2006 19:21:57 -0500 From: Michael Graves [EMAIL PROTECTED] Subject: [Asterisk-Users] New SkypeSIP gateway To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Anyone seen or tried this yet? http://www.voip-weblog.com/50226711/uplink_connects_sip_skype.php Michael - I have tried to register with both Asterisk and SER; Unfortunately this does not seem to work. Great idea. Guess we need to wait for the next version. I'll post some comments to the nch website. Shad ___ --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] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)
On 4/5/06, Matt Florell [EMAIL PROTECTED] wrote: In my experience capacity is a huge problem. You can't have sphinx running on 48 channels at once. It is limited to only a few instances at a time. Although I only did trials with sphinx2. What version are you using? and what dictionary? Sphinx2 - A customized dictionary. What would happen if you tried to run it on 48 channels at once? Is it a server issue? ___ --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] Phones are all auto answering
On Tue, 2006-04-04 at 10:44 -0400, Christian Buchter wrote: Strange, but all the phones when called immediately return a user is on the phone and the phone never rings. Anyone else ever experience this before? TIA Have the users managed to set DND on the phones? That would give the exact symptom. Rgds Pete ___ --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] Loading module chan_zap.so failed! PLZ help me!
I have recompiled my zaptel drivers but I still get the same error --- Derek Whitten [EMAIL PROTECTED] a écrit : ali asma wrote: I modified the configuration but I still have the same error. Please tell me in whach directory should I execute: modprobe zaptel modprobe wcfxo becose it seems that my card not had been detected Thanks, --- Lee Archer [EMAIL PROTECTED] a écrit : I run suse 10 and have an X100p. But I use fxsks=1 in the /etc/zaptel.conf not /etc/asterisk/zaptel.conf. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ali asma Sent: 04 April 2006 10:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Loading module chan_zap.so failed! PLZ help me! Hi, Sorry my card is X101P. My config is : /etc/asterisk/zaptel.conf : loadzone=us defaultzone=us fxoks=1 and /etc/asterisk/zapata.conf : [trunkgroups] [channels] context=mainmenu signalling=fxo_ks faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 language=en channel=1 please help me --- ali asma [EMAIL PROTECTED] a écrit : Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration == Channel map: 0 channels configured. and when I run asterisk it shows me this: Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_zap.so]Apr 4 09:45:58 WARNING[9975]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Apr 4 09:45:58 WARNING[9975]: loader.c:499 load_modules: Loading module chan_zap.so failed! Where do i look, how can i debug? Thanks in advance, ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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 ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --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 ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. === message truncated === ___ --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 ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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] Phones are all auto answering
Snom 190s and 220s, it seems to happen intermittently but not sure why -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, April 04, 2006 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phones are all auto answering What phones you using? On 4/4/06, Christian Buchter [EMAIL PROTECTED] wrote: Strange, but all the phones when called immediately return a user is on the phone and the phone never rings. Anyone else ever experience this before? TIA _ This email has been scanned by MessageLabs on behalf of E-INS ___ --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 _ This email has been scanned by MessageLabs on behalf of E-INS _ This email has been scanned by MessageLabs on behalf of E-INS ___ --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] Phones are all auto answering
What phones you using? On 4/4/06, Christian Buchter [EMAIL PROTECTED] wrote: Strange, but all the phones when called immediately return a user is on the phone and the phone never rings. Anyone else ever experience this before? TIA _ This email has been scanned by MessageLabs on behalf of E-INS ___ --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] Realtime Database Lookup
Hi,
Please take a look at the following extensions.conf:-
exten = _11,1,NoCDR()exten = _11,2,Dial(SIP/${EXTEN},10)exten = _11,3,VoiceMail()
I'm already using realtime for some extensions/users/voicemail.
Is there any way to do the following at point 3?:-
Lookup the realtime users db and read the MailBox column for that buddy.
If the mailbox column is empty, play a message saying Sorry, no one is available.
If the column has data in it, do the following:-
exten = _11,3,VoiceMail(MailBoxID)
Many thanks
Dan Journo
http://www.TextOver.com
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[Asterisk-Users] chan_modem_i4l delay
Hi, I currently use Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig of memory When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1 minutes , 2 sec after 3 minutes and so on... After a quart hour, the delay make the conversation just impossible !!! I use a tdm400P to connect my analogs phones and all is working very well between two zap stations. I have tried different Passive isdn card ( no hfc so I can't use zaphfc driver) Anybody have an idea to fix this problem ? BTW, I have compiled my kernel with the dtmf patch for isdn_tty.c so The cpu usage is 25% during a conversation, 75% idle I have a PCI latency of 32 msec With or without APIC, no changes It seems that the voice is buffered and sended too slowly to the i4l channel and so a delay is present afetr a short time and became bigger minutes after minutes... Alain Degreffe ___ --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] MeetMe/Asterisk Timer
Kelvin Williams wrote: We are using Asterisk in a purely VOIP environment, on leased dedicated server at a dedicated server provider. It is becoming more and more apparent that this dedicated server is actually a vritualized server. We have now found a need to utilize the MeetMe application for conferencing. However we have no Zaptel hardware. We have attempted to build the ztdummy kernel module for the server but are finding ourselves unable to do so because we do not have the kernel source on the box (as the dedicated server provider does not make it available, and typical resources for kernel sources causes the dedicated server to crash). In short, does anyone have any other advice to get the MeetMe application working on a potentially virtualized server (although the box said dedicated), without kernel sources, and a box that has no apparent USB? Thank you so much in advance for your advice. kw ___ --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 get your own hardware on a colo? signature.asc Description: OpenPGP digital signature ___ --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] New SkypeSIP gateway
Shad Mortazavi wrote: Message: 24 Date: Mon, 03 Apr 2006 19:21:57 -0500 From: Michael Graves [EMAIL PROTECTED] Subject: [Asterisk-Users] New SkypeSIP gateway To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Anyone seen or tried this yet? http://www.voip-weblog.com/50226711/uplink_connects_sip_skype.php Michael - I have tried to register with both Asterisk and SER; Unfortunately this does not seem to work. Great idea. Guess we need to wait for the next version. I'll post some comments to the nch website. Shad ___ --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 dam.. i was hoping for something server side, not some windoze client.. oh well.. guess it's back to more waiting for * - skype signature.asc Description: OpenPGP digital signature ___ --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] Frustrated with echo...
For phones, I've got a GS 101, a Sipura 841, and two analog phones hooked to an GS386 ATA (one phone per port). My troubles seem to be regardless of which phone is used, so I dont think it's on the phone-end of asterisk, but rather where I interface w/ Vonage and Verizon via POTS FXO... My SIP connections to the outside world have so far been good [frantically knocking on wood] I did go ahead and order the digium card yesterday evening, so I'm hopeful this will help. I had played with the gain, and was able to discern a difference, but it seemed to make some scenarios better, while making others worse, so I'm hoping the real digium card/drivers will just be smarter about handling it dynamically. Of course, my wife, who's a stay-at-home-mom is the biggest user of the system, but she's not interested in being a techy, so getting her to interrogate all callers about which number they dialed, etc.. and logging her opinions of the quality of the call hasn't worked! ;-) I also have some Cisco phones, but I haven't configured SCCP on my system yet, and dont want to use SIP on these phone (mostly to force myself to learn to configure SCCP on *) so that's another aspect that may help me after this weekend! Good point about the interrupts - I dont know the answer to that, but hopefully that'll also be a non-issue after I get the new card, and therefore have only one PCI slot handling everything. Thanks for the ideas!! -Steve From: Mike Dent [mailto:[EMAIL PROTECTED] Sent: Mon 4/3/2006 3:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Frustrated with echo... On 4/3/06, Steve Jones [EMAIL PROTECTED] wrote: I've been using my Asterisk (At my house - 2 modem-type fxos, and an assortment of SIP endpoints for phones) for about 5 weeks now, and I've been really happy with it, but I'm still having an echo problem that I've exhausted google with, and can't get straight... I think I've determined that because I'm using $7 voice modem clones for my FXOs that bad echo is going to just keep being a pain to me... I think I should have only tried going through proof of concept state with them, switching to something a little better quality when it was time to actually commit to Asterisk. So, my question is What's better and why: 1: a 'real' digium PCI card with two fxo plugins, or using a couple external SIP fxo units like a grandstream, zoom, or similar Personally, I think it would be desirable to keep the FXOs out of the asterisk box itself, just to give me future flexability to move to whatever the platform of the day I want to put asterisk on, without dealing with a PCI card to move, but if the consensus is that the voice quality and support for the digium board is the best, then that's what I'll do.. So, any comments on relative quality of these devices, and/or ones I've missed? 1: Grandstream HT-488 2: Zoom 5801/5802 3: DGM-TDM02B (TDM 400P with two FXOs) Are there any IAX2 FXOs that I'm missing? That seems to be an area that's oddly not taken care of... Any hints would be greatly appreciated! Steve, I have a similar setup at home, although I am in the UK. I've got the echo fairly well under control, however it seems much less when using my Cisco 7960 rather than the Grandstrean BT102 phone. Have you tried dropping the gain? Have you made sure you have both cards on seperate IRQ's which are not in use by network, video etc? I disabled USB and on board audio in the BIOS to help free up IRQ's. I think your best option is the TDM400 card, or perhaps consider the Sangoma card with a dual FXO module, maybe slightly cheaper! I'd be interested what SIP phones you are using and if echo differs between them. Mike winmail.dat___ --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] Fax over 2 bridged TE110P channels
To me, your * config files look correct. At a guess I'd say the problem is in
your motherboard. It is a sis chipset and from the look of things a couple
years old. Try running the system on an intel chipset motherboard and see how
you go.
Alternately, if you are running X windows, then disable that and see if it
improves things.
Craig
Alessio Focardi [EMAIL PROTECTED] wrote:
Hi,
I have an asterisk installation with 2 E1 cards
Software version is
Asterisk 1.2.6
Libpri 1.2.2
Zaptel 1.2.5
I'm having problem with fax transmission, let me explain better my
setup:
My fist TE110P E1 card is connected to the telco line
the second TE110P E1 one to an Nexspan PBX
so the server is basically sitting between the line, and the pbx.
every call coming from the line is simply redialed in the pbx
every call from pbx is simply redialed to the line
no answer is done
All is working great with voice, but faxing often results in error, both
receiving and sending.
I have disabled echo cancel, and also checked for interrupts problems
and other common misconfiguration problems.
Would someone please help me sort this out ?
I'm suspecting sync problems ...
Tnx for any help!
Following are some debug and config files
zaptel.conf
loadzone = it
defaultzone = it
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,0,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
zapata.conf
[channels]
switchtype = euroisdn
;line
signalling=pri_cpe
pridialplan=unknown
switchtype=euroisdn
priindication = outofband
echocancel=no
overlapdial=yes
immediate=no
nationalprefix=
internationalprefix=
resetinterval=300
context=pri1
group=1
channel = 1-15
channel = 17-31
;pbx
signalling=pri_net
pridialplan=international
switchtype=euroisdn
priindication=outofband
echocancel=no
overlapdial=yes
immediate=no
nationalprefix=
internationalprefix=
resetinterval=300
context=pri2
group=2
channel = 32-46
channel = 48-62
pri1 context
exten=_X.,1,Dial(Zap/g2/${EXTEN}||j)
exten=_X.,2,Congestion()
exten=_X.,102,Busy()
pri2 context
exten=_X.,1,Dial(Zap/g1/${EXTEN}||j)
exten=_X.,2,Congestion()
exten=_X.,102,Busy()
cat /proc/interrupts
CPU0
0: 1114420235 XT-PIC timer
1: 8 XT-PIC i8042
2: 0 XT-PIC cascade
5: 1114083499 XT-PIC t1xxp
8: 1 XT-PIC rtc
9: 0 XT-PIC acpi
10:2531734 XT-PIC eth0
12: 1114121836 XT-PIC t1xxp
14: 306435 XT-PIC ide0
NMI: 0
lspci -v
00:00.0 Host bridge: Silicon Integrated Systems [SiS] SiS645 Host Memory
AGP Controller (rev 01)
Flags: bus master, medium devsel, latency 32
Memory at e000 (32-bit, non-prefetchable) [size=64M]
Capabilities: [c0] AGP version 2.0
00:01.0 PCI bridge: Silicon Integrated Systems [SiS] Virtual PCI-to-PCI
bridge (AGP) (prog-if 00 [Normal decode])
Flags: bus master, fast devsel, latency 64
Bus: primary=00, secondary=01, subordinate=01, sec-latency=0
Memory behind bridge: dde0-dfef
Prefetchable memory behind bridge: d9c0-ddcf
00:02.0 ISA bridge: Silicon Integrated Systems [SiS] SiS961 [MuTIOL Media
IO]
Flags: bus master, medium devsel, latency 0
00:02.1 SMBus: Silicon Integrated Systems [SiS] SiS961/2 SMBus Controller
Flags: medium devsel
I/O ports at 0c00 [size=32]
00:02.5 IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] (rev d0)
(prog-if 80 [Master])
Subsystem: Silicon Integrated Systems [SiS] SiS5513 EIDE Controller
(A,B step)
Flags: bus master, fast devsel, latency 128
I/O ports at ff00 [size=16]
00:03.0 Ethernet controller: Silicon Integrated Systems [SiS] SiS900 PCI
Fast Ethernet (rev 90)
Subsystem: Silicon Integrated Systems [SiS] SiS900 10/100 Ethernet
Adapter
Flags: bus master, medium devsel, latency 64, IRQ 10
I/O ports at dc00 [size=256]
Memory at dfffc000 (32-bit, non-prefetchable) [size=4K]
Expansion ROM at dffa [disabled] [size=128K]
Capabilities: [40] Power Management version 2
00:08.0 ISDN controller: Cologne Chip Designs GmbH: Unknown device 16b8 (rev
01)
Subsystem: Cologne Chip Designs GmbH: Unknown device b562
Flags: medium devsel, IRQ 11
I/O ports at d800 [size=8]
Memory at d000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
00:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
Subsystem: Unknown device 6159:0001
Flags: bus master, medium devsel, latency 64, IRQ 5
I/O ports at d400 [size=256]
Memory at dfffe000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power
Re: [Asterisk-Users] IAX: Auto-congesting call due to slow response
maybe firewall tends to close iax connection, you can try to decrease qualify check interval (maybe qualify=5000?) PJ Mimmus wrote: Pavel Jezek wrote: I have same problem, do you have asterisk box behind nat? No, they are not behind NAT, peraphs there is a Checkpoint firewall. Bob McDowell wrote: It's been a while, but I didn't think those two terms were necessarily exclusive. Checkpoint firewalls can provide NAT, can they not? No, no! In this case I'm sure there is no NAT. Some other idea? Thanks Domenico ___ --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] Phones are all auto answering
Strange, but all the phones when called immediately return a user is on the phone and the phone never rings. Anyone else ever experience this before? TIA _ This email has been scanned by MessageLabs on behalf of E-INS ___ --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] Possible PRI fault?
On Tuesday 04 April 2006 10:39, Lee Archer wrote: I've been looking through the logs of a system trying to figure out why it sometimes starts extra asterisk processes. In the logs I keep seeing Define starts extra asterisk processes. Apr 4 15:22:18 WARNING[5054] chan_zap.c: Can't fix up channel from 1 to 2 because 2 is already in use Apr 4 15:22:18 WARNING[5054] chan_zap.c: Unable to move channel 2! Apr 4 15:22:55 WARNING[5054] chan_zap.c: Can't fix up channel from 1 to 4 because 4 is already in use This sounds like the telco is trying to specify which B channel to use. My understanding is that Asterisk does not currently support this. Asterisk chooses the B channel for outgoing calls. Did it ever work, or is this a new problem? -A. ___ --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] How to check if a phone / line is used?
ChanIsAvail allows you to see if a channel can *accept* calls, not if it is
currently in use. Here is a script that will fix you up:
checkchannel.agi - returns number of channels in use on a SIP peer
Sets a variable in the dialplan, MYCHANNELS, indicating number of channels
in use
#!/bin/bash
#Connect to the Asterisk console and dump a SIP SHOW CHANNELS command to
grep
#and filter out everything except the peer we are looking for
CHANNEL=`asterisk -rx SIP SHOW CHANNELS | grep -a -A0 .201`#Replace
.201 with the IP address of your SIP peer
#In this example, we have 4 registrations to the peer, you can carve out
unnessisary logic
#Initialize variables - here we are cutting out parts of the output to
create the variables
#You may have to change the location of the cut in order to make it work for
your install - this is for 1.0.9
CURRENTCHANNEL1=${CHANNEL:55:7}
CURRENTCHANNEL2=${CHANNEL:118:7}
CURRENTCHANNEL3=${CHANNEL:181:7}
CURRENTCHANNEL4=${CHANNEL:244:7}
TOTALCHANNELS=0
#If channel 1 is not an empty string and the string equals the ulaw codec,
it must be in use
#therefore increment the TOTALCHANNELS variable
#Replace the string 'ulaw' with the expected codec
#Optionally you could search for the string 'unknown' in order to determine
that a channel is NOT in use
if [ ${CHANNEL:55:7} != ]
then
if [ $CURRENTCHANNEL1 = ulaw ]
then
TOTALCHANNELS=$((TOTALCHANNELS+1))
fi
fi
#And so on - for 1 channel only, delete these 3 other if-fi's
if [ ${CHANNEL:118:7} != ]
then
if [ $CURRENTCHANNEL2 = ulaw ]
then
TOTALCHANNELS=$((TOTALCHANNELS+1))
fi
fi
if [ ${CHANNEL:181:7} != ]
then
if [ $CURRENTCHANNEL3 = ulaw ]
then
TOTALCHANNELS=$((TOTALCHANNELS+1))
fi
fi
if [ ${CHANNEL:244:7} != ]
then
if [ $CURRENTCHANNEL4 = ulaw ]
then
TOTALCHANNELS=$((TOTALCHANNELS+1))
fi
fi
#finally, dump a variable back to Asterisk indicating the number of channels
in use
echo SET VARIABLE MYCHANNELS \$TOTALCHANNELS\
hth
-Original Message-
From: Paul Zimm [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 05, 2006 8:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to check if a phone / line is used?
In the dialplan you can use ChanIsAvail command
Show channels?
On Mar 31, 2006, at 2:09 AM, Ronald Wiplinger wrote:
In the past I used SetGroup and CheckGroup to figure out if my
allowed providers lines are all used or not.
Since most of my provider have given me a single line anyway, I
wonder if there is a way to check if this (provider) line is taken
already.
How can I do that?
Same is with the phone. How can I see in CLI if a phone is now in use
or not?
Sip show peers shows me just if it is on-line, but not if it is in
a call or not.
In the dialplan I could dial the number and if it is busy, it would
go to the Voicemail for unavailable or busy. I expect that there is
just a test function as well, without trying to call.
bye
Ronald Wiplinger
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[Asterisk-Users] Possible PRI fault?
Title: Possible PRI fault? I've been looking through the logs of a system trying to figure out why it sometimes starts extra asterisk processes. In the logs I keep seeing Apr 4 15:22:18 WARNING[5054] chan_zap.c: Can't fix up channel from 1 to 2 because 2 is already in use Apr 4 15:22:18 WARNING[5054] chan_zap.c: Unable to move channel 2! Apr 4 15:22:55 WARNING[5054] chan_zap.c: Can't fix up channel from 1 to 4 because 4 is already in use Apr 4 15:22:55 WARNING[5054] chan_zap.c: Unable to move channel 4! Apr 4 15:26:49 WARNING[5054] chan_zap.c: Call specified, but not found? Apr 4 15:26:49 WARNING[5054] chan_zap.c: Unable to move channel 1! Apr 4 15:26:53 WARNING[5054] chan_zap.c: Call specified, but not found? Apr 4 15:26:53 WARNING[5054] chan_zap.c: Unable to move channel 2! Does this indicate a PRI problem? I am running with a TE110P card and I have identical systems running that don't have this problem. Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --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] long delay between Ring Begin and SIP/XXX is ringing
hi all, i have an asterisk install with a digium 4 port fxo card and cisco 7960 sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz 256KB cache and 1GB of ram. when a call comes in on zap/1-1 for example, the delay between when zap sees the line going to ring state, and when the desktop telephone rings can be as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece). below is some of the log -- note 2 seconds to get from Ring Begin to In Use and a total of 7 seconds before the sip phone rings. anyway to speed this process up? cheers charles Apr 5 16:15:00 DEBUG[7010]: chan_zap.c:6639 do_monitor: Monitor doohicky got event Ring Begin on channel 1 Apr 5 16:15:02 DEBUG[7010]: chan_zap.c:6639 do_monitor: Monitor doohicky got event Ring/Answered on channel 1 Apr 5 16:15:02 DEBUG[6986]: devicestate.c:187 do_state_change: Changing state for Zap/1 - state 2 (In use) Apr 5 16:15:02 DEBUG[7549]: app_queue.c:471 changethread: Device 'Zap/1' changed to state '2' (In use) -- Starting simple switch on 'Zap/1-1' Apr 5 16:15:05 NOTICE[7548]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... snip Apr 5 16:15:07 DEBUG[7012]: chan_sip.c:1447 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found -- SIP/101-7014 is ringing -- simplified chinese is not nearly as easy as they would have you believe ... a superlative oxymoron --anonymous ___ --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] Frustrated with echo...
Steve Jones wrote: I thought the whole thing with the hardware echo cancellation is that it was basically in liu of the equivilent echo cancellation done in software... The reason to go to the hardware was for high-density systems?? For two FXOs, I thought I'd be safe in getting the non-echo cancellation cards, but perhaps no, huh?! :-( The echo cancellation issue is highly dependent on the exact pstn lines that you use. The * software EC works well in lots of implementations, however it does have limits that seem to be directly related to the delay between the time data is sent verses when the reflected energy (echo) is received. The limit seems to be somewhere in the 30 to 35 millisecond range given the tests that I've conducted using various s/w tools. In very general terms, it seems the longer the pstn copper lines between asterisk and the Central Office, the more likely software EC will not be as usable or consistent as the hardware EC. The hardware EC (from digium or sangoma) have wider limits, and those limits are different for the digium TDM2400 verses sangoma A200D. The difference between the two cards is related to the exact hardware chipset used on the two cards. (The two chipsets have very different investment/engineering costs.) It should also be noted that many telephone companies have implemented various remote line concentrators (typically seen as relatively small steal boxes in the neighborhood) that can also have an impact on echo. There are no published guidelines or guesses that would suggest if the telco has implemented A then you must use EC B. Also note that whatever worked for cards and EC in one case is not at all indicative of what will work in another case as the pstn line construction is always different. (E.g., different length of copper cable, different gauge of cable, different methods of terminating the telco side of the pstn line, different quality of cables, different manufacturers and architectures of remote concentrators, etc, etc.) In all cases, regardless of whether one is using hardware or software EC, the efficiency of the EC function is highly dependent on transmission levels. If the transmission levels are set to high, echo is going to happen regardless of what card or EC is implemented. That seems to be an issue that many asterisk newbies (as well as lots of asterisk s/w developers) do not seem to understand. So, if you were going to be selling asterisk boxes throughout your region, one might consider having an arsenal of analog products that can be selected based on each specific implementation. For short pstn lines, the TDM400 card with s/w EC seems to work well for lots of folks. For longer pstn lines where echo is not properly addressed in s/w, the TDM2400 or A200D seems to address the problem. For long pstn lines and those that have somewhat unusual echo problems, the A200D seems to address more issues then what the TDM2400 does. The above does not address analog fax support, which also enters into the engineering decision. The choice is not necessarily one of supporting digium or not; its rather an engineering decision to select the product that addresses the technical issue, period. Unfortunately, there is no reasonable way for you (or anyone else) to know in advance which product is needed to address the issue. Anyone that tries to influence you otherwise is absolutely full of BS. That's based on 20+ years doing detailed engineering work (including pbx transmission engineering) for a very large US telco, AND, been-there-done-that with asterisk over a two to three year period. ___ --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_modem_i4l delay
I had the same problem with i4l. It seems to be a driver problem. I think i4l is depricated for a reason in the newer Asterisk versions. Funny thing is: When I switch the remote users into a MeetMe room. And have the local users dial in to the same meetme room. Then the problem disappears (at least for me). I don't know how or why this is. But it is my experience. I am not using i4l anymore. Rene Kluwen Chimit -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Alain Degreffe Sent: woensdag 5 april 2006 17:18 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] chan_modem_i4l delay Hi, I currently use Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig of memory When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1 minutes , 2 sec after 3 minutes and so on... After a quart hour, the delay make the conversation just impossible !!! I use a tdm400P to connect my analogs phones and all is working very well between two zap stations. I have tried different Passive isdn card ( no hfc so I can't use zaphfc driver) Anybody have an idea to fix this problem ? BTW, I have compiled my kernel with the dtmf patch for isdn_tty.c so The cpu usage is 25% during a conversation, 75% idle I have a PCI latency of 32 msec With or without APIC, no changes It seems that the voice is buffered and sended too slowly to the i4l channel and so a delay is present afetr a short time and became bigger minutes after minutes... Alain Degreffe ___ --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] Problem with setting ringtones on Cisco 7960 phone.
I am having this exact same problem. I have tried 7.5, 7.4, and 8.2. I have tried setting ALERT_INFO and _ALERT_INFO and have tried several ringtones without any luck. According to the WIKI, it should work: [snippet] Controlling ring tones from Asterisk By setting the Asterisk variable ALERT_INFO before you call Dial, Asterisk will add ringer tone info to the SIP invite that is sent to the phone. exten = 3010,1,SetVar(ALERT_INFO=Bellcore-dr1) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) Note: In SIP_HEAD or v1+ you wil need to do the following: exten = 3010,1,SetVar(_ALERT_INFO=something) Available ring tones by default Bellcore-BusyVerify Bellcore-Stutter Bellcore-MsgWaiting Bellcore-dr1 Bellcore-dr2 Bellcore-dr3 Bellcore-dr4 Bellcore-dr5 [end snippet] Does anybody have any ideas? Thanks! On Thu, 30 Mar 2006, Greg Mudd wrote: Hi All, I am running into a problem setting the ringtones via _ALERT_INFO on the Cisco 7960 phone. I am using * 1.2.1 and have tried setting the variable to several values. I have also tried setting the phone's software to both 7.5 and 8.2 thinking that it might be a version issue, but with no success. I have examined the packets and do see the ALERT_INFO header being sent, but the phone is not responding. Thanks in advance for any help you can provide. ~Greg ___ --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] VPB cannot call out
Check your DTMF Settings. --- hensem boy [EMAIL PROTECTED] wrote: Hi all I have a problem when I want to call out using VPB trunk line, it cannot send the DTMF. Is there anyone has the same problem? Please share with me the solution. Thanks. - New Yahoo! Messenger with Voice. Call regular phones from your PC and save big. ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)
The load on the system will crash your server with that many instances of real-time sphinx running. Take a look at 'top' while you run it on tow channels at once an see what the load is. MATT--- On 4/5/06, Matt [EMAIL PROTECTED] wrote: On 4/5/06, Matt Florell [EMAIL PROTECTED] wrote: In my experience capacity is a huge problem. You can't have sphinx running on 48 channels at once. It is limited to only a few instances at a time. Although I only did trials with sphinx2. What version are you using? and what dictionary? Sphinx2 - A customized dictionary. What would happen if you tried to run it on 48 channels at once? Is it a server issue? ___ --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] GoDaddy royally screws over aussievoip.com.au and soft-swtich.org
That is why I back up my web server to an ftp server in a diffrent data center :) --- Rob Thomas [EMAIL PROTECTED] wrote: Well, I wake up this morning, and aussievoip isn't up. I ring godaddy, who _were_ hosting it, and they say that the machine's been compromised, and you can't have your data. Nyah Nyah. I spent 1 hour and 38 minutes on the phone to them, trying to convince them to let me somehow get access to it, but to no avail. I've reported it to the Australian Federal Police High-Tech Crime Unit, asking for a forensic analysis of the attack - hopefully I'll be able to get a copy of the data from the police, eventually, that way. Until then, however, we're out of luck. I've had a couple of offers of hosting (I put it on voip-info.org) but for the moment, I've signed up with serverpronto, which does get 1440 hits from google on 'serverpronto sucks'- which is an order of magnitude less than 'godaddy sucks', at 155,000 hits. (With quotes, it gets 3 hits, and godaddy gets 783) So, basically, aussievoip.com and soft-switch.org will be down for AT LEAST 24 hours. I've spoken to coppice and he has a reasonably recent backup, but I'll be crawling google's cache for anything in there to try to rebuild aussievoip. Yes, I had backups. They were on the machine. It was a shared hosting server. You'd expect never to have data _loss_, just fumble-finger-ism. Obviously, I was wrong. GoDaddy sucks, indeed. Anyway, it's being taken care of, just don't expect there to be any aussievoip for at least a couple of days. --Rob ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Patch 5779 on 1.0.9?
oej's MeterMaid patch for monitoring parked calls through hints: http://bugs.digium.com/view.php?id=5779 Anyone tried it on 1.0.9? ___ --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] can't start chan_capi with asterisk group
Hello, While upgrading * from 1.0.9 to 1.2.5, I have installed chan-capi-head and I cant start asterisk under asterisk group asterisk -gc -U asterisk and asterisk -gc -U asterisk -G dialout work well but asterisk -gc -U asterisk -G asterisk fail. I am thinking about a group permission configuration but I have exactly the same one than with my old 1.0.9 working config. Log messages when launching asterisk -gc -U asterisk -G asterisk : Apr 5 17:47:21 VERBOSE[5773] logger.c: [chan_capi.so]Apr 5 17:47:21 VERBOSE[5773] logger.c: [chan_capi.so] = (Common ISDN API for Asterisk) Apr 5 17:47:21 VERBOSE[5773] logger.c: == Parsing '/etc/asterisk/capi.conf': Apr 5 17:47:21 VERBOSE[5773] logger.c: == Parsing '/etc/asterisk/capi.conf': Found Apr 5 17:47:21 WARNING[5773] chan_capi.c: CAPI not installed, CAPI disabled! Apr 5 17:47:21 WARNING[5773] loader.c: chan_capi.so: load_module failed, returning -1 Apr 5 17:47:21 WARNING[5773] loader.c: Loading module chan_capi.so failed! Ls l /dev/capi20 : crw-rw 1 root dialout 68, 0 2006-03-24 14:49 /dev/capi20 id asterisk : uid=105(asterisk) gid=105(asterisk) groupes=105(asterisk),20(dialout),33(www-data) Any idea about why I cant start chan_capi with asterisk group? ___ --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] long delay between Ring Begin and SIP/XXX is ringing
i have an asterisk install with a digium 4 port fxo card and cisco 7960 sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz 256KB cache and 1GB of ram. when a call comes in on zap/1-1 for example, the delay between when zap sees the line going to ring state, and when the desktop telephone rings can be as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece). below is some of the log -- note 2 seconds to get from Ring Begin to In Use and a total of 7 seconds before the sip phone rings. anyway to speed this process up? cheers charles Apr 5 16:15:00 DEBUG[7010]: chan_zap.c:6639 do_monitor: Monitor doohicky got event Ring Begin on channel 1 Apr 5 16:15:02 DEBUG[7010]: chan_zap.c:6639 do_monitor: Monitor doohicky got event Ring/Answered on channel 1 Apr 5 16:15:02 DEBUG[6986]: devicestate.c:187 do_state_change: Changing state for Zap/1 - state 2 (In use) Apr 5 16:15:02 DEBUG[7549]: app_queue.c:471 changethread: Device 'Zap/1' changed to state '2' (In use) -- Starting simple switch on 'Zap/1-1' Apr 5 16:15:05 NOTICE[7548]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... snip Apr 5 16:15:07 DEBUG[7012]: chan_sip.c:1447 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found -- SIP/101-7014 is ringing It would appear the progress is associated with waiting for callerid info. If you are in the US, callerid occurs between the first and second ring. That's about 7 seconds or so. If your pstn line does not have callerid, then add statements into your zapata.conf file like 'usecallerid=no', 'immediate=yes', etc. I don't recall exactly which statements are needed, but start with the above two and see what you get for delays. ___ --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_modem_i4l delay
On Wed, 5 Apr 2006, Alain Degreffe wrote: Hi, I currently use Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig of memory When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1 minutes , 2 sec after 3 minutes and so on... After a quart hour, the delay make the conversation just impossible !!! I use a tdm400P to connect my analogs phones and all is working very well between two zap stations. I have tried different Passive isdn card ( no hfc so I can't use zaphfc driver) Anybody have an idea to fix this problem ? What card is that? Why don't you use mISDN? Armin___ --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] SIP Asterisk Polycom Reinvite
Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are canreinvite=yes and the PSTN termination path is canreinvite=no then calls between polycoms should not have asterisk in the media stream and wan link utilization is reduced. The problem looks like the Polycom keeps trying to reinvite the sonus and the call never sets up right, and not with all calls Any experience with this? Maybe there is a totally different issue I am overlooking? About 3 to 5% of all Polycom to PSTN via asteriskSIP peer calls are impacted. I have not set the Polycom canreinvite=no yet, hoping to not have to do that as the wan link is a t1 that is also used for data. Thanks for any help! Damon ___ --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] queue issue
On 04/05/06 21:37 Dov Bigio said the following: - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf on a related note, we notice that if we've set atxfer = *1 in features.conf and blindxfer=#1, then attended transfers dont work. somehow, the Queue app captures the '*' and hangs up the call. is this the behaviour others have observed ? obviously, since we've used *2 for auto monitor, that doesnt work as well. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] fax server functionality on Asterisk
List, how can I put fax server functionality on Asterisk? * as a reliable fax server for 500-1000 fax/day (mostly incoming)? Fax server should be like HylaFax, i.e. stable, low maintenance and functionality like receiving fax as email with PDF attachment, sending faxes per WHFC. Faxing with spandsp using bri_stuff (BeroNet/Junghanns quadBRI ISDN cards) shortens some faxes, or faxes loose lines, or when sending faxes a bri channel stays open for days (seems to be a sync problem). Any experiences/hints/suggestions? Or how would I best use Asterisk and Hylafax? Would IAXmodem work reliable? Anyone here using this with BRI? Or what about using Asterisk+HylaFax+CAPI (e.g. AVM C4)? Is there another way of using Asterisk as a reliable faxserver with BRI? Frank -- LocaNet oHG - http://www.loca.net Lindemannstrasse 81, D-44137 Dortmund tel +49 231 91596-23, mobil +49 172 2120354 SIP:[EMAIL PROTECTED] ___ --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] RE: Milliwatt Test Number List
Well give it a day and I will reply to my own questions. I guess my friends are right that I do talk to myself :) Anyways, Sprint called back and according to their technician, "Oh, I'm sorry, it looks like we do have milliwatt test lines that support 1004 Hz or 1.004 kHz test tones" So for those of you that might need this info in the future here is the info I found. For Sprint Canada (now Rogers Telecom) NPA 905 - 905-290-0102 NPA 416 - 416-916-8102 NPA 647 - 647-430-0102 For Bell Canada Apparently Bell Canada reserves NPA-NXX-1185 for milliwatt in most exchanges. I was able to get it working in 2 of the exchanges that I needed it in, however a nice little old lady answered the phone in one of the others, so it appears that they are not 100% consistent on this, but try a few of the NXX in your local calling area and you should find one. I found the following that suited my purpose: Toronto - 416-494-1185 Toronto - 416-439-1185 Oshawa - 905-404-1185 Regards, Bill From: William M. Sandiford Sent: Wednesday, April 05, 2006 1:04 AMTo: asterisk-users@lists.digium.comSubject: Milliwatt Test Number List Hello: Does anyone know of a list of milliwatt test numbers for debugging echo? Specifically I am looking for a milliwatt test number in Canada, preferrably ina 416 or 905 NPA exchangedifferent carriers would also be niceie. Bell Canada, GT, Sprint (Now Rogers Telecom) I called Rogers NOC and asked them for the milliwatt test numberthey didn't even know what it wasso I got escalated to a technician and he tried claiming that they didn't have one (I find that really hard to believe) Any help would be appreciated. Bill ___ --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 setting ringtones on Cisco 7960 phone.
I am having this exact same problem. I have tried 7.5, 7.4, and 8.2. I have tried setting ALERT_INFO and _ALERT_INFO and have tried several ringtones without any luck. Using the current svn trunk, here is what works: exten = 3010,1,Set(_ALERT_INFO=bellcore-r3) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) The above causes a 7960 to ring with two short rings. As I recall from playing with the 7960 a long time ago, the phone only has limited number of ring tones installed. E.g., what works for a Sipura (as one example) will be different for the 7960. ___ --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] Master.csv Shell Script
Can you think of any reason that this would not pick up on times after call is placed, and then disconnected. I noticed that the time does not change on the call times after a call has been made. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Monday, March 27, 2006 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Master.csv Shell Script If you've got PHP installed, here's one I made for our office: http://horanappraisals.com/asterisk/total_account_codes/ Run it with no parameters to check Master.csv in the current directory, or pass the filename to parse as the first parameter. # ./total_account_codes /var/log/asterisk/cdr-csv/Master.csv test total is 310 seconds or 5.17 minutes or 0.09 hours total is 33130 seconds or 552.17 minutes or 9.2 hours # The second line totals all lines with no account code specified. hth moj Jeremy wrote: Im not looking for anything super detailed, just something to run through the master.csv file and give total time per account code. . . .does anyone out there have a script like this I could work from? ___ --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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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] TE110P errors
Hi All I have a TE110P card connected to a PRI line. In my zaptel.conf I have: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us and my zapata.conf is: [channels] context=inbound-pri switchtype = national pridialplan=unknown ;pridialplan=international signalling = pri_cpe callerid=asreceived busydetect=no usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=no threewaycalling=no echocancel=yes echocancelwhenbridged=no echotraining=yes group = 1 channel = 1-23 When I load the wcte11xp module, I get the following: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.5 Echo Canceller: KB1 ACPI: PCI Interrupt :03:00.0[A] - GSI 16 (level, low) - IRQ 169 Controller version: 24 Wrote '10' but read '0' Wrote '11' but read '1' Wrote '12' but read '2' Wrote '13' but read '3' Wrote '14' but read '4' Wrote '15' but read '5' Wrote '16' but read '6' Wrote '17' but read '7' Wrote '18' but read '8' Wrote '19' but read '9' Wrote '1a' but read 'a' Wrote '1b' but read 'b' Wrote '1c' but read 'c' Wrote '1d' but read 'd' Wrote '1e' but read 'e' Wrote '1f' but read 'f' Wrote '30' but read '20' Wrote '31' but read '21' Wrote '32' but read '22' Wrote '33' but read '23' Wrote '34' but read '24' Wrote '35' but read '25' Wrote '36' but read '26' Wrote '37' but read '27' Wrote '38' but read '28' Wrote '39' but read '29' Wrote '3a' but read '2a' Wrote '3b' but read '2b' Wrote '3c' but read '2c' Wrote '3d' but read '2d' Wrote '3e' but read '2e' Wrote '3f' but read '2f' Wrote '40' but read '0' Wrote '41' but read '1' Wrote '42' but read '2' Wrote '43' but read '3' Wrote '44' but read '4' Wrote '45' but read '5' Wrote '46' but read '6' Wrote '47' but read '7' Wrote '48' but read '8' Wrote '49' but read '9' Wrote '4a' but read 'a' Wrote '4b' but read 'b' Wrote '4c' but read 'c' Wrote '4d' but read 'd' Wrote '4e' but read 'e' Wrote '4f' but read 'f' Wrote '60' but read '20' Wrote '61' but read '21' Wrote '62' but read '22' Wrote '63' but read '23' Wrote '64' but read '24' Wrote '65' but read '25' Wrote '66' but read '26' Wrote '67' but read '27' Wrote '68' but read '28' Wrote '69' but read '29' Wrote '6a' but read '2a' Wrote '6b' but read '2b' Wrote '6c' but read '2c' Wrote '6d' but read '2d' Wrote '6e' but read '2e' Wrote '6f' but read '2f' Wrote '90' but read '80' Wrote '91' but read '81' Wrote '92' but read '82' Wrote '93' but read '83' Wrote '94' but read '84' Wrote '95' but read '85' Wrote '96' but read '86' Wrote '97' but read '87' Wrote '98' but read '88' Wrote '99' but read '89' Wrote '9a' but read '8a' Wrote '9b' but read '8b' Wrote '9c' but read '8c' Wrote '9d' but read '8d' Wrote '9e' but read '8e' Wrote '9f' but read '8f' Wrote 'b0' but read 'a0' Wrote 'b1' but read 'a1' Wrote 'b2' but read 'a2' Wrote 'b3' but read 'a3' Wrote 'b4' but read 'a4' Wrote 'b5' but read 'a5' Wrote 'b6' but read 'a6' Wrote 'b7' but read 'a7' Wrote 'b8' but read 'a8' Wrote 'b9' but read 'a9' Wrote 'ba' but read 'aa' Wrote 'bb' but read 'ab' Wrote 'bc' but read 'ac' Wrote 'bd' but read 'ad' Wrote 'be' but read 'ae' Wrote 'bf' but read 'af' Wrote 'c0' but read '80' Wrote 'c1' but read '81' Wrote 'c2' but read '82' Wrote 'c3' but read '83' Wrote 'c4' but read '84' Wrote 'c5' but read '85' Wrote 'c6' but read '86' Wrote 'c7' but read '87' Wrote 'c8' but read '88' Wrote 'c9' but read '89' Wrote 'ca' but read '8a' Wrote 'cb' but read '8b' Wrote 'cc' but read '8c' Wrote 'cd' but read '8d' Wrote 'ce' but read '8e' Wrote 'cf' but read '8f' Wrote 'e0' but read 'a0' Wrote 'e1' but read 'a1' Wrote 'e2' but read 'a2' Wrote 'e3' but read 'a3' Wrote 'e4' but read 'a4' Wrote 'e5' but read 'a5' Wrote 'e6' but read 'a6' Wrote 'e7' but read 'a7' Wrote 'e8' but read 'a8' Wrote 'e9' but read 'a9' Wrote 'ea' but read 'aa' Wrote 'eb' but read 'ab' Wrote 'ec' but read 'ac' Wrote 'ed' but read 'ad' Wrote 'ee' but read 'ae' Wrote 'ef' but read 'af' FALC version: TE110P: Setting up global serial parameters for T1 FALC V1.2 TE110P: Successfully initialized serial bus for card Found a Wildcard: Digium Wildcard TE110P T1/E1 Registered tone zone 0 (United States / North America) TE110P: Span configured for ESF/B8ZS Calling startup (flags is 4099) wcte1xxp: Setting yellow alarm I have tried compiling zaptel and libpri from the download source on asterisk.org, I have pulled them from svn, and I even tried [EMAIL PROTECTED], all had the same effect. I have also tried the card in three different systems. Is this a config issue that I am missing, or can I assume a bad card? Thanks, Kenny signature.asc Description: This is a digitally signed message part ___ --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] GoDaddy royally screws over aussievoip.com.au and soft-swtich.org
Well, I wake up this morning, and aussievoip isn't up. I ring godaddy, who _were_ hosting it, and they say that the machine's been compromised, and you can't have your data. Nyah Nyah. Have you tried the Internet archieve (wayback machine). I was once lucky to find my web pages there to recover it! I spent 1 hour and 38 minutes on the phone to them, trying to convince them to let me somehow get access to it, but to no avail. I've reported it to the Australian Federal Police High-Tech Crime Unit, asking for a forensic analysis of the attack - hopefully I'll be able to get a copy You are kidding, are you? You do not really expect that, do you? bye Ronald Wiplinger of the data from the police, eventually, that way. Until then, however, we're out of luck. I've had a couple of offers of hosting (I put it on voip-info.org) but for the moment, I've signed up with serverpronto, which does get 1440 hits from google on 'serverpronto sucks'- which is an order of magnitude less than 'godaddy sucks', at 155,000 hits. (With quotes, it gets 3 hits, and godaddy gets 783) So, basically, aussievoip.com and soft-switch.org will be down for AT LEAST 24 hours. I've spoken to coppice and he has a reasonably recent backup, but I'll be crawling google's cache for anything in there to try to rebuild aussievoip. Yes, I had backups. They were on the machine. It was a shared hosting server. You'd expect never to have data _loss_, just fumble-finger-ism. Obviously, I was wrong. GoDaddy sucks, indeed. Anyway, it's being taken care of, just don't expect there to be any aussievoip for at least a couple of days. --Rob ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. begin:vcard fn:Ronald Wiplinger n:Wiplinger;Ronald org:ELMIT Co., Ltd. adr:Shilin District;;5F., No.8, Alley 2, Lane 92, Dexing W. Road;Taipei;;11158;Taiwan email;internet:[EMAIL PROTECTED] title:CEO tel;work:+886.2.2835.7765 tel;cell:+886.939.775.516 x-mozilla-html:TRUE url:http://www.elmit.net version:2.1 end:vcard ___ --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_modem_i4l delay
My kernel is a 2.4.27 and I think that mISDN is available only for a 2.6.x But I can't use a 2.4.26 for some security reasons... Alain -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Armin Schindler Envoyé : mercredi 5 avril 2006 18:05 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] chan_modem_i4l delay What card is that? Why don't you use mISDN? Armin ___ --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] Need 25-50 Linksys boxes
Hi Andy - Anyone care to quote on 25 Linksys PAP2-NA units unlocked can email me direct. Straight forward sale best price new equip etc etc... I am a buyer located in the U.S. Need someone with stock that can ship right away. Will want 25 more in less than a week. You may get a better response if you post this to the biz list rather than the users list. Ideally the users list is not supposed to include commercial inquiries. - Noah ___ --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] SHOWCHANINFO Not Working
Hi,
SHOWCHANINFO outputs no data in the following line:-
exten = 1571,2,VoiceMailMain(${SIPCHANINFO(peername)[EMAIL PROTECTED])
So that command becomes:-
exten = 1571,2,VoiceMailMain(@incoming)
Can anyone help?
Thanks
Dan Journo
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Re: [Asterisk-Users] fax server functionality on Asterisk
Frank Ochmann wrote: List, how can I put fax server functionality on Asterisk? * as a reliable fax server for 500-1000 fax/day (mostly incoming)? Fax server should be like HylaFax, i.e. stable, low maintenance and functionality like receiving fax as email with PDF attachment, sending faxes per WHFC. Faxing with spandsp using bri_stuff (BeroNet/Junghanns quadBRI ISDN cards) shortens some faxes, or faxes loose lines, or when sending faxes a bri channel stays open for days (seems to be a sync problem). Any experiences/hints/suggestions? Or how would I best use Asterisk and Hylafax? Would IAXmodem work reliable? Anyone here using this with BRI? Or what about using Asterisk+HylaFax+CAPI (e.g. AVM C4)? Is there another way of using Asterisk as a reliable faxserver with BRI? Frank nvfaxdetect ? signature.asc Description: OpenPGP digital signature ___ --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] Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
Hi Marco My asterisk for all my users, everything was fine for 3 days, but now i can't access it. But it is running... Could any one help me on this? Can you provide some specific information? At least the following: Asterisk version Operating System Hardware Technologies used (zap, sip, etc) - Noah ___ --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] GoDaddy royally screws over aussievoip.com.au and soft-swtich.org
Unfortunetly, you learnt the hard way. Never rely on any third party. Make sure you have backups of all your data on machines which you have direct access to. There are a large number of offsite companies offering backup services. Checkthem out and make sure their contracts still allow you to sue them if they lose your data. We have two machines for just this reason. (We also learnt the hard way.)They mirror each other and they are hosted with two different companies. Therefore, if one company/server fails, we can instantly switch over without losing data. Never rely too heavily on one supplier/service. Good luck getting back online. Dan On 05/04/06, Ronald Wiplinger [EMAIL PROTECTED] wrote: Well, I wake up this morning, and aussievoip isn't up. I ring godaddy, who _were_ hosting it, and they say that the machine's been compromised, and you can't have your data. Nyah Nyah.Have you tried the Internet archieve (wayback machine). I was once lucky to find my web pages there to recover it! I spent 1 hour and 38 minutes on the phone to them, trying to convince them to let me somehow get access to it, but to no avail. I've reported it to the Australian Federal Police High-Tech Crime Unit, asking for a forensic analysis of the attack - hopefully I'll be able to get a copyYou are kidding, are you? You do not really expect that, do you? byeRonald Wiplinger of the data from the police, eventually, that way. Until then, however, we're out of luck. I've had a couple of offers of hosting (I put it on voip-info.org) but for the moment, I've signed up with serverpronto, which does get 1440 hits from google on 'serverpronto sucks'- which is an order of magnitude less than 'godaddy sucks', at 155,000 hits. (With quotes, it gets 3 hits, and godaddy gets 783) So, basically, aussievoip.com and soft-switch.org will be down for AT LEAST 24 hours. I've spoken to coppice and he has a reasonably recent backup, but I'll be crawling google's cache for anything in there to try to rebuild aussievoip. Yes, I had backups. They were on the machine. It was a shared hosting server. You'd expect never to have data _loss_, just fumble-finger-ism. Obviously, I was wrong. GoDaddy sucks, indeed. Anyway, it's being taken care of, just don't expect there to be any aussievoip for at least a couple of days. --Rob ___ --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 __ Do You Yahoo!? Tired of spam?Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 --Ronald Wiplinger(CEO of ELMIT)http://www.elmit.comhttp://voip.elmit.com http://e-paper.elmit.comTel. (M) +886.939.775.516(O) +886.2.2835.7765 (ENUM) or FWD 511208- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention!Our system is protected with a spam prevention program.If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please.After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] RE: Milliwatt Test Number List
Any clue for other countries (western Europe, for example) ?Cheers ___ --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] Re: Asterisk start/stop
change asterisk.conf: mkdir /var/run/asterisk chown it to your asterisk user. change astrundir = /var/run to astrundir = /var/run/asterisk My guess would be that you are running asterisk as a non-root user and that this user can not write to /var/run . if so, the ctl and PID files are not created. -- -- Steven http://www.glimasoutheast.org Tom Castleman [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi there, I have installed asterisk and freepbx on a Debian Sarge system. I followed the INSTALL doc and all is well apart from the starting and stopping of asterisk. I have linked /usr/sbin/amportal into rc2.d and all is well on initial startup, however when issuing an 'amportal stop' command as root I get Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) when it attempts to stop asterisk. Then I run amportal start and it says it is already running (obviously if it never stopped). Also when attempting to access the asterisk console as root, 'asterisk -rv' for example, I get the same message. I su to the asterisk user and get the same message. On a probably unrelated note, if I attempt to start asterisk via '/etc/init.d/asterisk start' (how I had it setup before installing freepbx) it starts then stops, exiting error code 1 or something, then starts and stops and starts etc etc. I think maybe I must need to slightly refined permissions on starting and stopping asterisk and locations of things. If any one could offer any advice/help it would be most appreciated. Kind regards, Tom Castleman. --- This SF.Net email is sponsored by xPML, a groundbreaking scripting language that extends applications into web and mobile media. Attend the live webcast and join the prime developer group breaking into this new coding territory! http://sel.as-us.falkag.net/sel?cmd=lnkkid=110944bid=241720dat=121642 ___ --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_modem_i4l delay
OOps The correct answer is My kernel is a 2.4.27 and I think that mISDN is available only for a 2.6.x But I can't use a 2.6.x for some security reasons... Alain -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL PROTECTED] Envoyé : mercredi 5 avril 2006 18:35 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE: [Asterisk-Users] chan_modem_i4l delay My kernel is a 2.4.27 and I think that mISDN is available only for a 2.6.x But I can't use a 2.4.26 for some security reasons... Alain -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Armin Schindler Envoyé : mercredi 5 avril 2006 18:05 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] chan_modem_i4l delay What card is that? Why don't you use mISDN? Armin ___ --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] Favorite softphone with command line interface
Hello,Which is your favorite SIP softphone with command line interface (ie with text imputs and outputs along with graphical GUI) ?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] Master.csv Shell Script
I run it, make a call, and after the call disconnects, when I run the
script again, I do get changed numbers:
[EMAIL PROTECTED] ~]$ total_account_codes /var/log/asterisk/cdr-csv/Master.csv
total is 151974 seconds or 2532.9 minutes or 42.22 hours
[EMAIL PROTECTED] ~]$ pbxmonitor
Mojo 7478633
[EMAIL PROTECTED] ~]$ total_account_codes /var/log/asterisk/cdr-csv/Master.csv
total is 151983 seconds or 2533.05 minutes or 42.22 hours
[EMAIL PROTECTED] ~]$
Is this what you mean? As you can see, I even allow blank accountcodes
('cause we don't even use accountcodes), so there shouldn't be any
issue. Maybe * is not logging the call duration under certain
disconnect circumstances.
As per the line near the top of my script, if I change the 12 to a 13 to
capture BillSeconds instead of duration, I get the following:
[EMAIL PROTECTED] ~]$ total_account_codes /var/log/asterisk/cdr-csv/Master.csv
total is 151639 seconds or 2527.32 minutes or 42.12 hours
which is slightly less than before... You might try this and see if it
helps.
Moj
Jeremy wrote:
Can you think of any reason that this would not pick up on times after call
is placed, and then disconnected. I noticed that the time does not change on
the call times after a call has been made.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: Monday, March 27, 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Master.csv Shell Script
If you've got PHP installed, here's one I made for our office:
http://horanappraisals.com/asterisk/total_account_codes/
Run it with no parameters to check Master.csv in the current directory, or
pass the filename to parse as the first parameter.
# ./total_account_codes /var/log/asterisk/cdr-csv/Master.csv
test total is 310 seconds or 5.17 minutes or 0.09 hours total is 33130
seconds or 552.17 minutes or 9.2 hours
#
The second line totals all lines with no account code specified.
hth moj
Jeremy wrote:
Im not looking for anything super detailed, just something to run
through the master.csv file and give total time per account code. . .
.does anyone out there have a script like this I could work from?
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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan Company, LLC
(907) 747- x112
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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] Fax over 2 bridged TE110P channels
2006/4/4, Remco Barende [EMAIL PROTECTED]: I suspect that in your case the fax channels are not natively bridged. I'mnot sure whether native bridging will work if you are using 2 cards.How would you prove that native bridging works (I mean independantly of current server processor or PCI bus load) ? cheers ___ --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] Asterisk support to Tornado M5 IP Phones
Hi there, Anyone knows the Tornado M5 IP Phones? I need to connect them to Asterisk, but I could not found any info. Best regards, Ing. Juan Carlos Huerta Director de Desarrollo Nucleum, la voz de tu red [EMAIL PROTECTED] www.nucleum.com.mx ___ --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] legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI) and callerid
Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID I am able only set callerid number (and name) but on phone is always only callerid number... 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] long delay between Ring Begin and SIP/XXX is ringing
On Wed, 2006-04-05 at 11:02 -0500, Rich Adamson wrote: snip It would appear the progress is associated with waiting for callerid info. If you are in the US, callerid occurs between the first and second ring. That's about 7 seconds or so. If your pstn line does not have callerid, then add statements into your zapata.conf file like 'usecallerid=no', 'immediate=yes', etc. I don't recall exactly which statements are needed, but start with the above two and see what you get for delays. 1 second :-) thanks a million! charles ___ --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] Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
I've been told that the problem was: I've a daily cron job: /usr/sbin/asterisk -r -x stop when convenient then i had /usr/sbin/asterisk start I've been recomended to replace: /usr/sbin/safe_asterisk I've done that, let's see how it goes tomorrow when i arrive at the office. I didn't have time yet to understand the safe_asterisk, if any one could summarize it would be very good Thanks, Best regards, Marco Mouta On 4/5/06, Noah Miller [EMAIL PROTECTED] wrote: Hi Marco My asterisk for all my users, everything was fine for 3 days, but now i can't access it. But it is running... Could any one help me on this? Can you provide some specific information? At least the following: Asterisk version Operating System Hardware Technologies used (zap, sip, etc) - Noah ___ --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] one-waysilence during calls
Title: Messaggio My sip phones are connected to asterisk PBX 1.2.4. The PBX is connected to the provider through IAX2 connection. Sometimes randomly the voice is stopped and both caller and called don't hear the other's voice. During this silence period Asterisk is not logging any errors. This happen on incoming calls too, and incoming calls are through ISDN BRI lines Any idea? 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
[Asterisk-Users] SIP client looses register and then i need to restart my pc to get registered on Asterisk 1.2.5
Hi all, I've a some users on my network, reporting this: Sjphone is registered , and some times just looses registry in Asterisk, I don't know if it is expiration ( instead of loosing registry). Then to get registered again they need to restart their own PC. Why could this beeing happening? Best regards, Marco Mouta ___ --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] can't start chan_capi with asterisk group
It should work with that permissions. Does it work with other group/user settings? Just for a try, set /dev/capi20 to rw-rw-rw Armin On Wed, 5 Apr 2006, amaury BOSSE wrote: Hello, While upgrading * from 1.0.9 to 1.2.5, I have installed chan-capi-head and I can't start asterisk under asterisk group asterisk -gc -U asterisk and asterisk -gc -U asterisk -G dialout work well but asterisk -gc -U asterisk -G asterisk fail. I am thinking about a group permission configuration but I have exactly the same one than with my old 1.0.9 working config. Log messages when launching asterisk -gc -U asterisk -G asterisk : Apr 5 17:47:21 VERBOSE[5773] logger.c: [chan_capi.so]Apr 5 17:47:21 VERBOSE[5773] logger.c: [chan_capi.so] = (Common ISDN API for Asterisk) Apr 5 17:47:21 VERBOSE[5773] logger.c: == Parsing '/etc/asterisk/capi.conf': Apr 5 17:47:21 VERBOSE[5773] logger.c: == Parsing '/etc/asterisk/capi.conf': Found Apr 5 17:47:21 WARNING[5773] chan_capi.c: CAPI not installed, CAPI disabled! Apr 5 17:47:21 WARNING[5773] loader.c: chan_capi.so: load_module failed, returning -1 Apr 5 17:47:21 WARNING[5773] loader.c: Loading module chan_capi.so failed! Ls -l /dev/capi20 : crw-rw 1 root dialout 68, 0 2006-03-24 14:49 /dev/capi20 id asterisk : uid=105(asterisk) gid=105(asterisk) groupes=105(asterisk),20(dialout),33(www-data) Any idea about why I can't start chan_capi with asterisk group? ___ --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 setting ringtones on Cisco 7960 phone.
I suppose that works. I get two short rings. Is there a way to change the actual sound of it, though? On Wed, 5 Apr 2006, Rich Adamson wrote: I am having this exact same problem. I have tried 7.5, 7.4, and 8.2. I have tried setting ALERT_INFO and _ALERT_INFO and have tried several ringtones without any luck. Using the current svn trunk, here is what works: exten = 3010,1,Set(_ALERT_INFO=bellcore-r3) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) The above causes a 7960 to ring with two short rings. As I recall from playing with the 7960 a long time ago, the phone only has limited number of ring tones installed. E.g., what works for a Sipura (as one example) will be different for the 7960. ___ --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] legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI) and callerid
Miroslav HOSTINSKY napisał(a): Hello, I have connected asterisk box with legacy PBX Alcatel OmniPCX 4400 (and also another * box connected to A4200). These PBXes have function to assign name to extensions and display it on phone. Yes. They do :-D. A4400, current amount: 3. A4220E currently only one in storage room. And plenty of new model: OXE. Asterisk box is connected via PRI with euroISDN signalling (also I have tried QSIG). iirc EuroISDN requires PRA/PRA2/BRA2, the qSIG requires DLT. Is it possible to set callerid with name and display it on alcatel digital phones? With command SetCALLERID I am able only set callerid number (and name) but on phone is always only callerid number... Yes, and no. On PRI you could have only number. The string is not going enywhere. I suppose that QSIG mode should be the answer. Do you set, clid as 'someone 1234' ? kd, -- Krzysztof Drewicz Affordable 2/4 span E1/T1 PCI-cards. 100% Asterisk compatible. See http://4e1.pl ___ --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] Asterisk svn starting problem
5 apr 2006 kl. 08.52 skrev René Enskat [Teamware GmbH]: hi i updated asterisk today via svn no i can'T start asterisk i get core dumps. i have to comment some modules then i can start: noload = format_au.so noload = format_mp3.so noload = format_pcm_alaw.so.so noload = format_pcm_alaw.so We changed the interface for format drivers, so old drivers can't load. Some drivers was integrated into others, and the Makefile needs to remove these. At this point, you need to read the warning and delete them manually. Sorry for the trouble. format_mp3 will hopefully be fixed in asterisk-addons soon. Oh, life in the development version - svn trunk. It's risky, but fun! /Olle PS. Continue to test the test branch! We all need your help. While testing you can listen to this message from our project founder: http://svn.digium.com/view/asterisk/tags/0.1.3/sounds/demo- moreinfo.gsm --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk European Tour: http://www.meetasterisk.com ___ --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] Asterisk on BSD?
The subject says it all I think. I'm looking at maybe needing to run it under BSD 5 Thanks in advance Bruce ___ --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] fax server functionality on Asterisk
Frank Ochmann escreveu: how can I put fax server functionality on Asterisk? * as a reliable fax server for 500-1000 fax/day (mostly incoming)? Fax server should be like HylaFax, i.e. stable, low maintenance and functionality like receiving fax as email with PDF attachment, sending faxes per WHFC. I use app_rxfax for incoming fax, it works well. -- Paulo ___ --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] zaphfc NT Mode. Extension not recognized...
Hi all I finaly set up a second * with two ZapHFC Cards. One in TE the other in NT mode. So I have a 1.2.5 Asterisk to run Meetme etc... and a 1.2.4 Asterisk to run all that Zaptel stuff. First I used mISDN on 1.2.5 which worked, but sometimes had strange behaviour. So my hope was that zaptel is more stable... It steams realy stable in TE mode. No problem there. But I get a very strange behaviour in NT mode. I have two different ISDN phones here. Booth worked with mISDN. Only one can dial with zaphfc. With the other * does not recognize the dialed extension... Here is what I get: Phone 1 (working one): I press 11 and take it of hook: == Primary D-Channel on span 1 up for TEI 64 -- Extension '11' in context 'from-zap' from '0010618115711' does not exist. This is fine. I dialed 11 and it actuely does not exist. Phone 2 (the one which is unable to dial) I press 11 and then take it off-hook: == Primary D-Channel on span 1 up for TEI 64 -- Accepting voice call from '010618115711' to 's' on channel 0/2, span 1 -- Executing NoOp(Zap/2-1, BLAH s) in new stack I seam not to find any way to get the dialed extension passed from the phone to *. What am I doing wrong? I did try DISA to check if the number is passed after the phone is off hook, but this just doesn't seam the case... The extension was recognized with mISDN. Any ideas? -Benoit- ___ --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] Asterisk RealTime queue - periodic-announce
Hi List, is there a reason why Asterisk Realtime queues don't support periodic_announce_frequency and periodic_announce options? I have tried adding the 2 fields to my MySQL table, but they seem to be ignored? Any hints are appreciated. Regards Kristian ___ --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] Asterisk on BSD?
On 11:12, Wed 05 Apr 06, Bruce Ferrell wrote: The subject says it all I think. I'm looking at maybe needing to run it under BSD 5 It runs fine on OpenBSD 3.8 No zaptel though, but for FreeBSD there's a zaptel port. http://ezine.daemonnews.org/200409/asterisk.html http://www.voip-info.org/tiki-index.php?page=Asterisk+FreeBSD Have fun -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] The Asterisk bug tracker :: please think twice before opening a report!
Friends, At this point, we're close to 300 issues open in the bug tracker at http://bugs.digium.com Some of us spend many hours each week, if not each day, to work with the bug tracker. It's a tool for us, a very important tool to handle new features and find bugs in Asterisk, tracking them down. It is important that you consider a few things while using this tool: - If a bug marshal closes your report, do not re-open it. If you want to discuss the action, use the #asterisk-dev IRC channel or the asterisk-dev mailing list. If you continuosly re-open a bug report we close, you will get a karma reduction. Other bug marshals may disagree and re-open the bug report, but don't do it yourself if a bug marshal directed you to take the report or discussion somewhere else. You will only annoy the bug marshals, doing no good for you or your cause. - The bug tracker is not a support forum. Do not open a bug report to ask a question. Do not open a bug report to get help with your configuration. The asterisk-users mailing list and the #asterisk IRC channel are good places for this, as is the Asterisk web based forum. - If you open a bug report, make sure you respond quickly. Most of the people working on the bug tracker are contributing their own time to work on the issues. Make sure that you assist them while they are working to solve your issues. If you do not respond, the report will be closed regardless of the importance. We simply can't proceed without your feedback. - Feature requests is better discussed on the mailing lists. If you open a feature request on the bug tracker, it will be kept open for a few days. After that, it will be closed but not erased from the database. It's still reachable, but not in the list of open issues. Feature requests in the bug tracker seldom lead to new code. Better to find a developer, pay for new code and contribute that. IF YOU REALLY HAVE A BUG OR A NEW FEATURE Please do use the bug tracker to report bugs! Don't be scared of the amount of open bug reports. If it is a bug, it is a bug and we need the report. First, remember to - Read the bug guidelines and follow them - Try to locate an existing bug report for your issue Please help us testing new features, please help us locating bugs. See if you can make a bug reported in the bug tracker to show up on your system and report that fact. Some bugs are hard to find, and we need help finding out if a bug can be repeated or not. Thank you for your assistance and understanding! Asterisk bug marshals and developers through /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)
It wacked up to maybe 20% for all of 300ms while it was processing the data from the caller... hrmmm On 4/5/06, Matt Florell [EMAIL PROTECTED] wrote: The load on the system will crash your server with that many instances of real-time sphinx running. Take a look at 'top' while you run it on tow channels at once an see what the load is. MATT--- On 4/5/06, Matt [EMAIL PROTECTED] wrote: On 4/5/06, Matt Florell [EMAIL PROTECTED] wrote: In my experience capacity is a huge problem. You can't have sphinx running on 48 channels at once. It is limited to only a few instances at a time. Although I only did trials with sphinx2. What version are you using? and what dictionary? Sphinx2 - A customized dictionary. What would happen if you tried to run it on 48 channels at once? Is it a server issue? ___ --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 ___ --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] Queues - Dumb question
- Original Message - From: Wes Baehr [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, April 03, 2006 3:16 PM Subject: [Asterisk-Users] Queues - Dumb question It was my understanding that when an agent answers a queue call, he will not be hit with another call until he finishes his current call. Currently, my agents get hit with calls from the queue while they are still on a previous call, so I've resorted to setting their call-limit in sip.conf to 1. But, this prevents them from putting one call on hold and making another call (although they could use parking). Maybe I misunderstood, but I'm asking anyway :) Wes, I don't know that ours is the best solution, but we addressed this by turning call waiting off on our agents SIP phones. We use Snom 320s and there is a Call Waiting Indicator under the advanced section. Like you I originally set the call limit to 1, but that is not feasable because our agents have to place outgoing calls and transfer the caller, and we can't have additional calls incomming during this time. On the bright side one thing I noticed is that if the agent never picks up the second call, the caller remains in the queue, though while they are ringing the second line they can't get answered by a different agent until Asterisk decides to give up on that agent and try another. I tested this by being the caller and confirmed I never heard anything but music on hold even while the second line on the agent's phone was flashing, though I cannot swear you are getting the same behaviour with your phones. Oh and I also had an issue with Snom Firmware higher than 4.5 that caused this same problem, so we've had to stay with the 4.5 firmware. Hope this is of some help, Frank Webb InterMedia Marketing Solutions Assistant Project Leader ___ --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] transforming g729 files to wav files
Hello list, is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php Tofik Suleymanov ___ --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] SIP T
5 apr 2006 kl. 16.40 skrev Jon Weisman: Anyone know how I can get SIP T working w/ Asterisk? Start with explaining your definition of SIP T then we can look into it :-) /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] SIP T
Well what I need is to get the info digits on a sip call (toll free orignation) and send that call out a PRI to my PSTN switch via FeatureGroupD so that I know where the call is originating from. Can I do this with Asterisk? And how??? -Jon - Original Message - From: Olle E Johansson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 05, 2006 3:37 PM Subject: Re: [Asterisk-Users] SIP T 5 apr 2006 kl. 16.40 skrev Jon Weisman: Anyone know how I can get SIP T working w/ Asterisk? Start with explaining your definition of SIP T then we can look into it :-) /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] IAX2 Origination Problem
Hi all, I have here several IAX2 Softphones(IDEFISK, DIAX and an own develop based on iaxclient.lib). I have follow dialrules in my std-test extension: [std-test] exten = *601,1,Answer exten = *601,n,Dial(IAX2/pbxnetwork/xx,30,m) exten = *601,n,Hangup exten = *602,1,Answer exten = *602,n,Dial(IAX2/pbxnetwork/xx,30) exten = *602,n,Hangup No I have a problem when I use the Manager API and the Originate Action, when I Originate my Softphone to *601 I hear the holding-music and get connected right to the opposite site. That works since January without problems and everything is fine. But when I originate to *602 I hear nothing. I see in the CLI that the Call get connected but I not hear anything in my Softphone. One curios thing is that in IDEFISK I get a the opposite site when I click on the Active Line Selection, but this is onliest client where I have seen this behaviour. My Originate statement is: Action: Originate Channel: IAX2/test Exten: *602 Callerid: IAX2/test Account: test Context: std-test Priority: 1 The Debug Outputs are: Output of Origination to *601 - Call accepted by xx.xxx.xxx.xx (format gsm) -- Format for call is gsm Channel IAX2/test-2 was answered. -- Executing Answer(IAX2/test-2, ) in new stack -- Executing Dial(IAX2/test-2, IAX2/pbxnetwork/0xx|30|m) in new stack -- Called pbxnetwork/0xx -- Started music on hold, class 'default', on channel 'IAX2/test-2' -- Call accepted by xx.xxx.xxx.xx (format gsm) -- Format for call is gsm -- IAX2/pbxnetwork-6 is making progress passing it to IAX2/test-2 -- IAX2/pbxnetwork-6 is ringing -- IAX2/pbxnetwork-6 answered IAX2/test-2 -- Stopped music on hold on IAX2/test-2 -- Hungup 'IAX2/pbxnetwork-6' -- Hungup 'IAX2/test-2' Output of Origination to *602 - Call accepted by xx.xxx.xxx.xx (format gsm) -- Format for call is gsm Channel IAX2/test-3 was answered. -- Executing Answer(IAX2/test-3, ) in new stack -- Executing Dial(IAX2/test-3, IAX2/pbxnetwork/0xx|30) in new stack -- Called pbxnetwork/0xx -- Call accepted by xx.xxx.xxx.xx (format gsm) -- Format for call is gsm -- IAX2/pbxnetwork-4 is making progress passing it to IAX2/test-3 -- IAX2/pbxnetwork-4 is ringing -- IAX2/pbxnetwork-4 stopped sounds -- IAX2/pbxnetwork-4 answered IAX2/test-3 -- Hungup 'IAX2/pbxnetwork-4' -- Hungup 'IAX2/test-3' I hope someone can give me a hint what the problem is and how I could solve it. With greetings Jan Serve. ___ --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 setting ringtones on Cisco 7960 phone.
hello, maybe quite off topic, but is there any way, how to do some like: exten = 3010,2,Dial(SIP/3010/ALERT_INFO=normal_ringtoneSIP/3011/ALERT_INFO=beep_ringtone) so, ring on two lines concurently, but using two distinguish tones (eg. I would like to be informed, about incomming call for other phone, but only with beep tone on my phone) PJ Jeremy Koski wrote: I suppose that works. I get two short rings. Is there a way to change the actual sound of it, though? On Wed, 5 Apr 2006, Rich Adamson wrote: I am having this exact same problem. I have tried 7.5, 7.4, and 8.2. I have tried setting ALERT_INFO and _ALERT_INFO and have tried several ringtones without any luck. Using the current svn trunk, here is what works: exten = 3010,1,Set(_ALERT_INFO=bellcore-r3) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) The above causes a 7960 to ring with two short rings. As I recall from playing with the 7960 a long time ago, the phone only has limited number of ring tones installed. E.g., what works for a Sipura (as one example) will be different for the 7960. ___ --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 ___ --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] IAX2 Origination Problem
CFN - Jan Serve wrote: Hi all, I have here several IAX2 Softphones(IDEFISK, DIAX and an own develop based on iaxclient.lib). I have follow dialrules in my std-test extension: [std-test] exten = *601,1,Answer exten = *601,n,Dial(IAX2/pbxnetwork/xx,30,m) exten = *601,n,Hangup exten = *602,1,Answer exten = *602,n,Dial(IAX2/pbxnetwork/xx,30) exten = *602,n,Hangup No I have a problem when I use the Manager API and the Originate Action, when I Originate my Softphone to *601 I hear the holding-music and get connected right to the opposite site. That works since January without problems and everything is fine. But when I originate to *602 I hear nothing. I see in the CLI that the Call get connected but I not hear anything in my Softphone. One curios thing is that in IDEFISK I get a the opposite site when I click on the Active Line Selection, but this is onliest client where I have seen this behaviour. My Originate statement is: Action: Originate Channel: IAX2/test Exten: *602 Callerid: IAX2/test Account: test Context: std-test Priority: 1 The Debug Outputs are: Output of Origination to *601 - Call accepted by xx.xxx.xxx.xx (format gsm) -- Format for call is gsm Channel IAX2/test-2 was answered. -- Executing Answer(IAX2/test-2, ) in new stack -- Executing Dial(IAX2/test-2, IAX2/pbxnetwork/0xx|30|m) in new stack -- Called pbxnetwork/0xx -- Started music on hold, class 'default', on channel 'IAX2/test-2' -- Call accepted by xx.xxx.xxx.xx (format gsm) -- Format for call is gsm -- IAX2/pbxnetwork-6 is making progress passing it to IAX2/test-2 -- IAX2/pbxnetwork-6 is ringing -- IAX2/pbxnetwork-6 answered IAX2/test-2 -- Stopped music on hold on IAX2/test-2 -- Hungup 'IAX2/pbxnetwork-6' -- Hungup 'IAX2/test-2' Output of Origination to *602 - Call accepted by xx.xxx.xxx.xx (format gsm) -- Format for call is gsm Channel IAX2/test-3 was answered. -- Executing Answer(IAX2/test-3, ) in new stack -- Executing Dial(IAX2/test-3, IAX2/pbxnetwork/0xx|30) in new stack -- Called pbxnetwork/0xx -- Call accepted by xx.xxx.xxx.xx (format gsm) -- Format for call is gsm -- IAX2/pbxnetwork-4 is making progress passing it to IAX2/test-3 -- IAX2/pbxnetwork-4 is ringing -- IAX2/pbxnetwork-4 stopped sounds -- IAX2/pbxnetwork-4 answered IAX2/test-3 -- Hungup 'IAX2/pbxnetwork-4' -- Hungup 'IAX2/test-3' I hope someone can give me a hint what the problem is and how I could solve it. With greetings Jan Serve. ___ --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 Can you do an iax2 debug to see if packets are travelling when you hear nothing? -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --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] transforming g729 files to wav files
Hi Tofik - is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php The wiki also lists GX::Transcoder which looks like it can do g729 to wav, though I've never tried it. Here's a link: http://www.germanixsoft.de/index.php Otherwise, you could probably rig up asterisk to transcode from g729 to another codec then record it to a file. There's probably not more tools to do this since most people aren't interested in going from the very lossy g729 codec to the non-lossy wav format. - Noah ___ --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] RE: Problem with setting ringtones on Cisco 7960 phone.
Is there an easy way to find out what ringtones a Cisco 79XX has installed? I've tried going through the Telnet interface, but can't find any lists of ringtones. Trying the code below produces a different kind of ring, but not two short rings as indicated. I've also seen the ringtone listed as Bellcore-dr3. Is it case sensitive? Thanks! I am having this exact same problem. I have tried 7.5, 7.4, and 8.2. I have tried setting ALERT_INFO and _ALERT_INFO and have tried several ringtones without any luck. Using the current svn trunk, here is what works: exten = 3010,1,Set(_ALERT_INFO=bellcore-r3) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) The above causes a 7960 to ring with two short rings. As I recall from playing with the 7960 a long time ago, the phone only has limited number of ring tones installed. E.g., what works for a Sipura (as one example) will be different for the 7960. ___ --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] Sending Access codes to a 5EE switch.
I have an Asterisk sever running with a TE406P card, and 4 pri T1s. I am trying to findout how to send access codes to the switch. After a long distance call is dialed, we get a second dial tone and I need to enter a 4 digit access code, then the switch will place the call. Does anyone know how I can do this? Or does anyone know how to tell asterisk to send to 4 digit code after it is dialed? Gary,, Gary Ritter, SCSA Network Technician Leaco Rural Telephone Coop. Inc. (505) 433-4326 office phone (505) 399-0062 cell phone [EMAIL PROTECTED] ___ --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] IAX2 Origination Problem
Joshua Colp wrote: Can you do an iax2 debug to see if packets are travelling when you hear nothing? Sure, but I not really can decrypt this: - Executing Dial(IAX2/test-6, IAX2/pbxnetwork/xx|30|tTr) in new stack -- Called pbxnetwork/xx Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00014ms SCall: 7 DCall: 0 [217.24.217.52:4569] VERSION : 2 CALLED NUMBER : xx CODEC_PREFS : () CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: IAX2/test LANGUAGE: en USERNAME: 109992 FORMAT : 2 CAPABILITY : 65283 ADSICPE : 0 DATE TIME : 2006-04-05 22:36:08 Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 005 Type: CONTROL Subclass: RINGING Timestamp: 04143ms SCall: 6 DCall: 03101 [84.188.169.95:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00016ms SCall: 00085 DCall: 7 [217.24.217.52:4569] AUTHMETHODS : 3 CHALLENGE : 160878529 USERNAME: 109992 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00021ms SCall: 7 DCall: 00085 [217.24.217.52:4569] MD5 RESULT : e94416602d7e3ea5be3f7ca3053dac18 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00021ms SCall: 00085 DCall: 7 [217.24.217.52:4569] FORMAT : 2 -- Call accepted by 217.24.217.52 (format gsm) -- Format for call is gsm Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00021ms SCall: 7 DCall: 00085 [217.24.217.52:4569] Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 04143ms SCall: 03101 DCall: 6 [84.188.169.95:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 03104 DCall: 0 [84.188.169.95:4569] USERNAME: test REFRESH : 60 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGACK Timestamp: 00012ms SCall: 8 DCall: 03104 [84.188.169.95:4569] USERNAME: test DATE TIME : 2006-04-05 22:36:10 REFRESH : 60 APPARENT ADDRES : IPV4 84.188.169.95:4569 Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 03104 DCall: 0 [84.188.169.95:4569] USERNAME: test REFRESH : 60 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 8 DCall: 03104 [84.188.169.95:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00012ms SCall: 03104 DCall: 8 [84.188.169.95:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: VOICE Subclass: 2 Timestamp: 01280ms SCall: 00085 DCall: 7 [217.24.217.52:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 01280ms SCall: 7 DCall: 00085 [217.24.217.52:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 002 Type: CONTROL Subclass: (14?) Timestamp: 01763ms SCall: 00085 DCall: 7 [217.24.217.52:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 01763ms SCall: 7 DCall: 00085 [217.24.217.52:4569] -- IAX2/pbxnetwork-7 is making progress passing it to IAX2/test-6 Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 002 Type: CONTROL Subclass: RINGING Timestamp: 04943ms SCall: 00085 DCall: 7 [217.24.217.52:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 04943ms SCall: 7 DCall: 00085 [217.24.217.52:4569] -- IAX2/pbxnetwork-7 is ringing Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 002 Type: CONTROL Subclass: (255?) Timestamp: 05663ms SCall: 00085 DCall: 7 [217.24.217.52:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 006 Type: IAX Subclass: ACK Timestamp: 05663ms SCall: 7 DCall: 00085 [217.24.217.52:4569] Rx-Frame Retry[ No] -- OSeqno: 006 ISeqno: 002 Type: CONTROL Subclass: ANSWER Timestamp: 05666ms SCall: 00085 DCall: 7 [217.24.217.52:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 007 Type: IAX Subclass: ACK Timestamp: 05666ms SCall: 7 DCall: 00085 [217.24.217.52:4569] -- IAX2/pbxnetwork-7 answered IAX2/test-6 Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 005 Type: CONTROL Subclass: (255?) Timestamp: 09793ms SCall: 6 DCall: 03101 [84.188.169.95:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 007 Type: VOICE Subclass: 2 Timestamp: 05660ms SCall: 7 DCall: 00085 [217.24.217.52:4569] Rx-Frame Retry[ No] -- OSeqno: 007 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 05660ms SCall: 00085 DCall: 7 [217.24.217.52:4569] Tx-Frame Retry[000] -- OSeqno: 005 ISeqno: 005 Type: IAX Subclass: LAGRQ Timestamp: 10017ms SCall: 6
RE: [Asterisk-Users] Pickup() h323
Jeremy McNamara wrote: Digium paid for ooh323, for whatever reasons that is beyond me, but it has proven to be no better than any H.323 channel driver, so I hope they got their money back. Better is subjective in this case. There's no doubt that chan_ooh323 has some warts. On the other hand it has NO external library requirements, and works out of the box with Cisco's Call Manager. One could argue that Call Manager is crap. Fine, that doesn't change the fact some of us are stuck with it. Chan_h323 did not work with CCM, and a query/bug report was dismissed, basically stating that Cisco was F'd up and the channel would not be updated to work with it unless funded. (fair, but not helpful) Chan_oh323 worked with CCM, but suffered from the external library requirements. Chan_ooh323 just worked. The code is, to a infrequent programmer, easy to read, extend and fix bugs. So for me chan_ooh323 is a 'better' H.323 channel driver. Dan ___ --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] Sending Access codes to a 5EE switch.
Gary,
What I do is the following:
In SIP.conf
Add the line accountcode= and set it equal to each users unique four digit
pin
example:
[user1]
secret=
accountcode=1234
type=friend
host=dynamic
context=default
canreinvite=no
nat=yes
qualify=2000
disallow=all
allow=g729
And in Extensions.conf
exten=_X.,1,Prefix(${ACCOUNTCODE})
exten=_X.,2,Dial,Zap/g1/${EXTEN}
-Jon
- Original Message -
From: Gary Ritter
To: asterisk-users@lists.digium.com
Sent: Wednesday, April 05, 2006 4:37 PM
Subject: [Asterisk-Users] Sending Access codes to a 5EE switch.
I have an Asterisk sever running with a TE406P card, and 4 pri T1's.
I am trying to findout how to send access codes to the switch. After a long
distance call is dialed, we get a second dial tone and I need to enter a 4
digit access code, then the switch will place the call. Does anyone know how
I can do this? Or does anyone know how to tell asterisk to send to 4 digit
code after it is dialed?
Gary,,
Gary Ritter, SCSA
Network Technician
Leaco Rural Telephone Coop. Inc.
(505) 433-4326 office phone
(505) 399-0062 cell phone
[EMAIL PROTECTED]
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Re: [Asterisk-Users] RE: Problem with setting ringtones on Cisco 7960 phone.
Well, if I look at my tftp directory where the phone downloads its config files, etc, on v7.1 I see a RINGLIST.DAT that contains the names of the ring files to be downloaded. On my system that includes ringer1.pcm and ringer2.pcm. I recall someone posting something about how to generate the content of the ringer1.pcm file, so my guess is that you can encode various sounds into such a file and call it via the _ALERT_INFO stuff shown. Some time ago, the following were valid ringtone names: ; Bellcore-BusyVerify ; Bellcore-Stutter ; Bellcore-MsgWaiting ; Bellcore-dr1 ; Bellcore-dr2 ; Bellcore-dr3 ; Bellcore-dr4 ; Bellcore-dr5 I don't have a clue if those names remain the same from one sip version to another; best guess is they do. Paul A. Pringle wrote: Is there an easy way to find out what ringtones a Cisco 79XX has installed? I've tried going through the Telnet interface, but can't find any lists of ringtones. Trying the code below produces a different kind of ring, but not two short rings as indicated. I've also seen the ringtone listed as Bellcore-dr3. Is it case sensitive? Thanks! I am having this exact same problem. I have tried 7.5, 7.4, and 8.2. I have tried setting ALERT_INFO and _ALERT_INFO and have tried several ringtones without any luck. Using the current svn trunk, here is what works: exten = 3010,1,Set(_ALERT_INFO=bellcore-r3) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) The above causes a 7960 to ring with two short rings. As I recall from playing with the 7960 a long time ago, the phone only has limited number of ring tones installed. E.g., what works for a Sipura (as one example) will be different for the 7960. ___ --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] Sending Access codes to a 5EE switch.
On Wednesday 05 April 2006 16:42, Jon Weisman wrote:
And in Extensions.conf
exten=_X.,1,Prefix(${ACCOUNTCODE})
exten=_X.,2,Dial,Zap/g1/${EXTEN}
That won't work for this case, as he needs to enter the access code *after*
dialing. Right offhand, I can't think of doing anything other than executing
a macro after dialing, and having the macro just SendDTMF() the access code.
-A.
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Re: [Asterisk-Users] transforming g729 files to wav files
The resulting file is not going to sound any better and its going to take up more space. What is the reason you need a WAV file? Perhaps there is a more efficient way to do what you are trying to do. Darrell S. Long BestWeb Corporation Tofik Suleymanov wrote: Hello list, is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php Tofik Suleymanov ___ --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] asterisk-ooh323, asterisk 1.2.6 and netmeeting
Dinesh Nair wrote: more tests reveal that with ohphone, calls from SIP-ohphone work fine with audio passed both ways. however when ohphone calls a SIP device, the call is hungup when the SIP device answers. This was sort of my problem too. I have two Asterisk servers, with an IAX2 trunk between them: Phone - Asterisk 1 - IAX - Asterisk 2 - H323 - Avaya IP403 - Phone If I dialled from a SIP phone on Asterisk 1 to the Phone on the Avaya, it worked fine. If I dialled from a phone on the Avaya, the SIP phone would ring, but the call would drop as soon as it was answered because of codec negotiation failure. After removing the various disallow= and allow= lines, the codec negotation is now successful in both directions. cYa, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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] How to restrict simultaneous phone registrations
Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? Thanks, Bryan Mahin Rediscover Personal Servicewith UNETA Please visit us @ www.uneta.com ___ --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] What does this error mean app.c: Huh....? no dial for indications?
Hi, What does the following error mean: Apr 5 12:39:40 NOTICE[22755] app.c: Huh? no dial for indications? Here is the 'full' log around the error: Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/[EMAIL PROTECTED],1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002 Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3001', on 'Local/[EMAIL PROTECTED],1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3001 Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|413|SIP/413) in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/413|20|Ttw) in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Called 413 Apr 5 12:38:24 VERBOSE[22758] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|510|SIP/510) in new stack Apr 5 12:38:24 VERBOSE[22758] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 12:38:24 VERBOSE[22758] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/510|20|Ttw) in new stack Apr 5 12:38:24 VERBOSE[22758] logger.c: -- Called 510 Apr 5 12:38:24 VERBOSE[22759] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|411|SIP/411) in new stack Apr 5 12:38:24 VERBOSE[22759] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 12:38:24 VERBOSE[22759] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/411|20|Ttw) in new stack Apr 5 12:38:24 VERBOSE[22759] logger.c: -- Called 411 Apr 5 12:38:24 VERBOSE[22758] logger.c: -- SIP/510-82b7 is ringing Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Agent/3002 is ringing Apr 5 12:38:25 VERBOSE[22759] logger.c: -- SIP/411-74d0 is ringing Apr 5 12:38:25 VERBOSE[22755] logger.c: -- Agent/3001 is ringing Apr 5 12:38:25 VERBOSE[22759] logger.c: -- SIP/411-74d0 is ringing Apr 5 12:38:25 VERBOSE[22757] logger.c: -- SIP/413-d3c8 is ringing Apr 5 12:38:25 VERBOSE[22755] logger.c: -- Agent/3005 is ringing Apr 5 12:38:29 VERBOSE[22758] logger.c: -- SIP/510-82b7 answered Local/[EMAIL PROTECTED],2 Apr 5 12:38:29 VERBOSE[22755] logger.c: -- Agent/3002 answered Zap/1-1 Apr 5 12:38:29 VERBOSE[22757] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 12:38:29 VERBOSE[22757] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 12:38:29 VERBOSE[22759] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 12:38:29 VERBOSE[22759] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 12:38:29 VERBOSE[22758] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 12:38:29 VERBOSE[22758] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 12:38:38 VERBOSE[22783] logger.c: -- Starting simple switch on 'Zap/3-1' Apr 5 12:39:39 VERBOSE[22755] logger.c: -- Started music on hold, class 'default', on Zap/1-1 Apr 5 12:39:39 VERBOSE[22755] logger.c: -- Playing 'pbx-transfer' (language 'en') Apr 5 12:39:40 NOTICE[22755] app.c: Huh? no dial for indications? Apr 5 12:39:42 VERBOSE[22755] logger.c: -- Stopped music on hold on Zap/1-1 Apr 5 12:39:42 VERBOSE[22755] logger.c: -- Executing Macro(Zap/1-1, stdexten|411|SIP/411) in new stack Apr 5 12:39:42 VERBOSE[22755] logger.c: -- Executing Set(Zap/1-1, DYNAMIC_FEATURES=automon) in new stack Apr 5 12:39:42 VERBOSE[22755] logger.c: -- Executing Dial(Zap/1-1, SIP/411|20|Ttw) in new stack Apr 5 12:39:42 VERBOSE[22755] logger.c: -- Called 411 Apr 5 12:39:44 VERBOSE[22755] logger.c: -- SIP/411-5f5d is ringing Apr 5 12:40:03 VERBOSE[22755] logger.c: -- Nobody picked up in 2 ms 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
[Asterisk-Users] What causes deadlock?
Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing Answer(Zap/5-1, ) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing SetMusicOnHold(Zap/5-1, default) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing DigitTimeout(Zap/5-1, 5) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Set Digit Timeout to 5 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing ResponseTimeout(Zap/5-1, 30) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Set Response Timeout to 30 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing GotoIfTime(Zap/5-1, 8:00-21:00|*|*|*?default|s|7) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Goto (default,s,7) Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing Queue(Zap/5-1, extensions-home|tr|||25) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- outgoing agentcall, to agent '3005', on 'Local/[EMAIL PROTECTED],1' Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Called Agent/3005 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/[EMAIL PROTECTED],1' Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Called Agent/3002 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- outgoing agentcall, to agent '3001', on 'Local/[EMAIL PROTECTED],1' Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Called Agent/3001 Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|413|SIP/413) in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/413|20|Ttw) in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Called 413 Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|510|SIP/510) in new stack Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/510|20|Ttw) in new stack Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Called 510 Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|411|SIP/411) in new stack Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/411|20|Ttw) in new stack Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Called 411 Apr 5 14:02:42 VERBOSE[23366] logger.c: -- SIP/510-1cb8 is ringing Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Agent/3002 is ringing Apr 5 14:02:43 VERBOSE[23365] logger.c: -- SIP/413-d49e is ringing Apr 5 14:02:43 VERBOSE[23363] logger.c: -- Agent/3005 is ringing Apr 5 14:02:43 VERBOSE[23365] logger.c: -- SIP/413-d49e is ringing Apr 5 14:02:43 VERBOSE[23367] logger.c: -- SIP/411-1a1a is ringing Apr 5 14:02:43 VERBOSE[23363] logger.c: -- Agent/3001 is ringing Apr 5 14:02:43 VERBOSE[23366] logger.c: -- SIP/510-1cb8 answered Local/[EMAIL PROTECTED],2 Apr 5 14:02:43 VERBOSE[23363] logger.c: -- Agent/3002 answered Zap/5-1 Apr 5 14:02:43 VERBOSE[23365] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 14:02:43 VERBOSE[23365] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 14:02:43 VERBOSE[23367] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 14:02:43 VERBOSE[23367] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 14:02:43 VERBOSE[23366] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 14:02:43 VERBOSE[23366] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Apr 5 14:03:22 VERBOSE[2424] logger.c: -- Registered SIP '412' at 10.0.0.68 port 5060 expires 120 Apr 5 14:05:35 VERBOSE[23363] logger.c: == Spawn extension (default, s, 7) exited non-zero on 'Zap/5-1' Apr 5 14:05:35 VERBOSE[23363]
[Asterisk-Users] cisco 7960
does one know how to program so i can have 2 lines on one sip account on that phone ?im runnign my own asteriskdo i need 2 local accounts ? one for each line ? that rebounds to same SIP forp VOIP provider ? ___ --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] WebMeetme Problem Please help!!!
Title: WebMeetme Problem Please help!!! I am running Feodra, I have downloaded the WebMeetMe Program, untar it to /var/www/html/WebMeetMe. I can access teh web page as of now. I cannot for the life of me figure out where defines.conf is. The install tells me it is in /var/www/html/WebMeetMe/lib/ however a complete search of the computer cannot find it anywhere. The /lib/ subdirectory does not exist in the untar'ed folder either. I could understand creating it under the /lib/ directory but I can't see a reason why it wouldn,t be there already. Here is what I have done... Download to /home/ directory extract to /var/www/html/ try to edit defines.php no directory or file Am I missing something crucial here? This is the directory of /var/www/html/WebMeetMe about.php conf_control.php css index.php phpagi_2_14 call_operator.php counter.txt images info.txt I checked the folders under this directory. All I can figure is that I am downloading 1.2 and the instructions are for 1.3. Which apparently is a bad link. ___ --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] Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4
I was just getting to work on fax for my * system, so I thought I would bring everything up to date since there would be some new compilations involved. yum update gave me kernel-2.6.16-1.2069_FC4 but after recompiling zaptel, I kept getting FATAL module zaptel not found Chased this for an hour with multiple recompiles and reboots. Finally dropped back to 2.6.15-1.1833_FC4, which worked before, and still works now. Bill ___ --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] How to restrict simultaneous phone registrations
Bryan Mahin wrote: Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? This is a non-issue, because a second registration to the same account will override and previous registrations for that account. ___ --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] Sending Access codes to a 5EE switch.
Andrew Kohlsmith wrote:
On Wednesday 05 April 2006 16:42, Jon Weisman wrote:
And in Extensions.conf
exten=_X.,1,Prefix(${ACCOUNTCODE})
exten=_X.,2,Dial,Zap/g1/${EXTEN}
That won't work for this case, as he needs to enter the access code *after*
dialing. Right offhand, I can't think of doing anything other than executing
a macro after dialing, and having the macro just SendDTMF() the access code.
show application dial Pay special attention to the D() option.
___
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RE: [Asterisk-Users] WebMeetme Problem Please help!!!
Title: WebMeetme Problem Please help!!! Sorry folks, my DSL took a bullet during a move this week and I'm still trying to get it back. Now I do see one problem, the correct file is defines.php not .conf. If my README file points to .conf, I will fix that (but from memory I don't think it does, so I wonder where it came from). Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan NovakSent: Wednesday, April 05, 2006 4:26 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] WebMeetme Problem Please help!!! I am running Feodra, I have downloaded the WebMeetMe Program, untar it to /var/www/html/WebMeetMe. I can access teh web page as of now. I cannot for the life of me figure out where defines.conf is. The install tells me it is in /var/www/html/WebMeetMe/lib/ however a complete search of the computer cannot find it anywhere. The /lib/ subdirectory does not exist in the untar'ed folder either. I could understand creating it under the /lib/ directory but I can't see a reason why it wouldn,t be there already.Here is what I have done...Download to /home/ directoryextract to /var/www/html/try to edit defines.phpno directory or fileAm I missing something crucial here?This is the directory of /var/www/html/WebMeetMeabout.php conf_control.php css index.php phpagi_2_14call_operator.php counter.txt images info.txtI checked the folders under this directory. All I can figure is that I am downloading 1.2 and the instructions are for 1.3. Which apparently is a bad link. ___ --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] cisco 7960
On Wed, 2006-04-05 at 17:54 -0400, Jimmy Smith wrote: does one know how to program so i can have 2 lines on one sip account on that phone ? im runnign my own asterisk do i need 2 local accounts ? one for each line ? that rebounds to same SIP forp VOIP provider ? Yes. ___ --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] cisco 7960
On Wed, 5 Apr 2006, Greg Oliver wrote: On Wed, 2006-04-05 at 17:54 -0400, Jimmy Smith wrote: does one know how to program so i can have 2 lines on one sip account on that phone ? im runnign my own asterisk do i need 2 local accounts ? one for each line ? that rebounds to same SIP forp VOIP provider ? Yes. The cisco phones can have multiple lines with the same registration... we had our phones set up like that until we decided to move to a one line call waiting type system. You just put the same account information in the configuration file for the second line as for the first line. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] ASTCC: How to reset in-use flag automatically ?
Ronald Wiplinger wrote: I tried now many places to put these lines in. The system still announces This card number is in use. Can you give me a place where to put it in? It's not receiving a card number. Find the following 3 lines: # # At this point we have a valid card number. # Insert the whole routine either just before or after these lines. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] How to restrict simultaneous phone registrations
Eric ManxPower Wieling wrote: Bryan Mahin wrote: Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? This is a non-issue, because a second registration to the same account will override and previous registrations for that account. While it is a non-issue, it is still annoying if both phones try to register all the time, . bye Ronald Wiplinger begin:vcard fn:Ronald Wiplinger n:Wiplinger;Ronald org:ELMIT Co., Ltd. adr:Shilin District;;5F., No.8, Alley 2, Lane 92, Dexing W. Road;Taipei;;11158;Taiwan email;internet:[EMAIL PROTECTED] title:CEO tel;work:+886.2.2835.7765 tel;cell:+886.939.775.516 x-mozilla-html:TRUE url:http://www.elmit.net version:2.1 end:vcard ___ --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] Setting ptime attribute in SDP invite
Is it possible for Asterisk to set the ptime attribute on outbound calls in SDP invite? ___ --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] Running into problems with the Digital Receptionist (Callers are not redirected to it)-
Hi- I'm a newbie to Asterisk, and in the process of setting up a working system. I'm kind of stuck with a problem regarding the Digital Receptionist, and I was hoping someone on this list might be able to shed some light on whats going on. So basically, I have the SIP phones/extensions and outbound trunks configured (I'm using Telasip trunks for outbound), and I am able to make and receive calls from my SIP phones to external cell phones. I have a 'front desk' extension configured to be 1998, and I have another two phones with extension 1901 and 1902 for my end users. I have a main DiD number through Telasip (lets call it 408-123-4567), and when that number is called, I want the following to happen: 1. Phone rings three times at extension 1998 (front desk) 2. If no one answers the phone, then the digital receptionist takes over and presents the caller with a menu (dial *411 for the company directory) 3. If the caller does not enter any extension, then the call goes to voicemail for extension 1998 This seems like a pretty straightforward setup, and I have seen many examples of this on the 'net, but unfortunately none of them work for me. Regardless of what I do, an external caller who dials my DiD (408-123-4567) goes through step 1, then straight to step 3 - completely skipping step 2. In other words, the digital receptionist is never called upon to present the menu. I have gone through the basic sanity checks: A. When I dial from an internal extension (1901), I get the digital receptionist and am presented with the option to dial the company directory. B. When I go to Amp - Setup - Incoming Calls and select 'Extension 1901' (or any extension, incl. 1998) instead of 'Digital Receptionist', callers from outside to the DiD are sent directly to the extension, and once again, steps 1 and 3 are executed in succession. It is almost as if I'm missing some line in extensions-custom.conf that tells Asterisk to invoke the digital receptionist, (possibly between lines 5 and 6? I'm just guessing here...) Here is what the extensions_custom.conf looks like: - [tsvxsj-in] exten = 4086241467,1,Answer exten = 4086241467,2,Wait(1) exten = 4086241467,3,Background(pls-hold-while-try) exten = 4086241467,4,NoOp(Incoming call on TelaSIP #4081234567) exten = 4086241467,5,Dial(SIP/1998,20,m) exten = 4086241467,6,Voicemail([EMAIL PROTECTED]) exten = 4086241467,7,Hangup - And here's what the debug log looks like when a call comes in from the outside, and Asterisk is set to send calls to the Digital Receptionist: - asterisk*CLI -- Executing Answer(SIP/telasip-username-e3f2, ) in new stack -- Executing Wait(SIP/telasip-username-e3f2, 1) in new stack -- Executing BackGround(SIP/telasip-username-e3f2, pls-hold-while-try) in new stack -- Playing 'pls-hold-while-try' (language 'en') -- Executing NoOp(SIP/telasip-username-e3f2, Incoming call for ArrayComm on TelaSIP #4081234567) in new stack -- Executing Dial(SIP/telasip-username-e3f2, SIP/1998|20|m) in new stack -- Called 1998 -- Started music on hold, class 'default', on channel 'SIP/telasip-username-e3f2' -- SIP/1998-f4fa is ringing -- Nobody picked up in 2 ms -- Stopped music on hold on SIP/telasip-username-e3f2 -- Executing VoiceMail(SIP/telasip-username-e3f2, [EMAIL PROTECTED]) in new stack -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/1998/INBOX/msg0001 format: wav49, 0x99fbe40 -- x=1, open writing: /var/spool/asterisk/voicemail/default/1998/INBOX/msg0001 format: wav, 0x9a002e0 -- User hung up -- Recording was 2 seconds long but needs to be at least 3 - abandoning == Spawn extension (tsvxsj-in, 4081234567, 6) exited non-zero on 'SIP/telasip-username-e3f2' asterisk*CLI - This seems like a pretty simple problem, and I've tried googling variants of 'Asterisk Digital Receptionist Now Working' and 'Asterisk Digital Receptionist Problem' with no results. I'm turning to you guys in the hope that someone will be able to tell me what I'm doing wrong. If there's anything else (configs, debug logs) that I need to post, just let me know and I'll do that as well. Thanks in advance- --Maxx ___ --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] Running into problems with the Digital Receptionist (Callers are not redirected to it)-
An obvious typo in here... Here is the corrected version: Here is what the extensions_custom.conf looks like: - [tsvxsj-in] exten = 4081234567,1,Answer exten = 4081234567,2,Wait(1) exten = 4081234567,3,Background(pls-hold-while-try) exten = 4081234567,4,NoOp(Incoming call on TelaSIP #4081234567) exten = 4081234567,5,Dial(SIP/1998,20,m) exten = 4081234567,6,Voicemail([EMAIL PROTECTED]) exten = 4081234567,7,Hangup - --Maxx ___ --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
