RE: [Asterisk-Users] 911 emergency service and VoIP
Joe, This is highly implementation specific. Perhaps I can give you some pointers to help you out. BTW, if you just happen to be in Texas, I can provide you with a list. Regular 911 calls are answered by a PSAP. Voip calls also goto a PSAP, but are handled differently. In fact, in most regions there aren't clear ways of handling these calls as of yet. Here are some pointers. 1. Do NOT call the PSAP. They are very busy, and in general are the WRONG organization to contact. Instead you want the 911 Agency for the area you are to serve. This is either a Council Of Governments or an Emergency Communications District depending on when it was formed. For example, here in Houston, the 911 Agency just happens to be the Harris County ECD. The Houston 911 Agency's website is (coincidential) http://www.911.org You must MUST start with them before you do ANYTHING 911. Certifications are required. http://www.nena.org is a good starting point.. Use search 2. If you are going to do 911, you must send LOCATION (ie: address) information to the 911 database. I do this through Intrado http://www.intrado.com through a product called data exchange 3. Depending on your connectivity, understand that the 911 agency and PSAP don't care what technology you use to connect to your customer. So if you can provide ANI, you are pretty much good to go. 4. For what it's worth; my traditional VoIP service offering will deliver 911 calls in the indentical manner as my non VoIP calls. If you'd like to talk specifics I can help you but I'd have to request that we take it off list since I feel that it is outside of the scope of the list. You can reach me at [EMAIL PROTECTED] -Brett -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Baptista Sent: Wednesday, June 16, 2004 8:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 911 emergency service and VoIP I understand that most VoIP providers allow for 911 calling but that 911 service is not the same as that available to PSTN. From what I understand a 911 Call Will Go To A General Access Line at the Public Safety Answering Point (PSAP). This is different from the 911 Emergency Response Center where traditional 911 calls go. Does anyone know how I can get information on howto contact the people at the Public Safety Answering Points (PSAPs)? Is there alist somewhere I can reference. thanks joe baptista ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 no compatible codecs
Hi All I have a strange problem using IAX2. When placing a call to my IAX clients (firefly) via the Asterisk dialplan all works great. However trying to initiate a call via the manager interface to the IAX client using the following command results in an error: Action: Originate Channel: IAX2/7000 Extension: 7000 Context: local Priority: 1 ActionID: 1 The error I get in the CLI is Jun 17 08:18:36 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by #IP: No compatible Codecs Does anyone have any ideas. Thanks in advance Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 no compatible codecs
check under your network settings that you have all the codecs selected and obviously type IAX Jason Penton wrote: Hi All I have a strange problem using IAX2. When placing a call to my IAX clients (firefly) via the Asterisk dialplan all works great. However trying to initiate a call via the manager interface to the IAX client using the following command results in an error: Action: Originate Channel: IAX2/7000 Extension: 7000 Context: local Priority: 1 ActionID: 1 The error I get in the CLI is Jun 17 08:18:36 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by #IP: No compatible Codecs Does anyone have any ideas. Thanks in advance Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modified Prepaid Error
hi, Am Do, den 17.06.2004 schrieb oi geli um 1:13: I am trying to install the Modified Prepaid App. I have installed PostgeSQL, created the tables, etc. Make Install runs ok. The when I try to launch asterisk (asterisk -vgc), it fails to run. I get the following errors, 1st error: [app_prepaid.so] = (Prepaid Application) == Parsing '/etc/asterisk/prepaid.conf': Found Jun 16 14:27:27 ERROR[-1085267840]: app_prepaid.c:127 check_connected: app_prepaid: cannot connect to database server localhost. Calls will not be logged == Registered application 'Prepaid' this simple means that the prepaid application can't connect to your postgres database. How does your prepaid.conf looks like ? - Have you also added the prepaid/asterisk user to your db ? best regards Wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no audio with sip
I can make call in
to the asterisk server listen to voice mail, and do the echo test. When make a
call I get no audio inbound or outbound. When making incoming call I can leave a
valid voice message, but when then extentions pick up again no audio inbound or
outbound.I am using Xten liteand Broadvoice. Below are the messages
from console when call is made and my sip.conf. Any
thoughts.
console
info:
-- Executing Dial("SIP/xlite2-725e", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/sip99413-6d20 is ringing -- SIP/sip99413-6d20
answered SIP/xlite2-725e -- Attempting native bridge of
SIP/xlite2-725e and SIP/sip99413-6d20 -- Attempting native
bridge of SIP/xlite2-725e and SIP/sip99413-6d20
sip.conf:
[general]disallow=allallow=ulawport=5060
; Port to bind
tobindaddr=0.0.0.0
; Address to bind SIP channel
toexternip=24.218.94.95localnet=192.168.2.0localmask=255.255.255.0context=default
; Default context for incoming
callsmaxexpirey=180defaultexpirey=160canreinvite=notos=reliabilitysrvlookup=yes
register =
4137711401:[EMAIL PROTECTED]/99413
[sip99413]secret=passwordusername=4137711401host=sip.broadvoice.comtype=friendnat=yescanreinvite=nodtmfmode=inbandfromuser=4137711401callerid=4137711401context=incomingfromdomain=sip.broadvoice.comqualify=yesdisallow=allallow=ulaw
[xlite2]type=friendusername=xlite2secret=passwordcallerid="outcast"
5678host=dynamicnat=yes
; X-Lite is behind a NAT
routercanreinvite=no
; Typically set to NO if behind
NATdisallow=allallow=ulaw
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004
RE: [Asterisk-Users] IAX2 no compatible codecs
Hi Adam Done all that but still the same problem. Do you have any other ideas? Cheers Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart Sent: 17 June 2004 08:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 no compatible codecs check under your network settings that you have all the codecs selected and obviously type IAX Jason Penton wrote: Hi All I have a strange problem using IAX2. When placing a call to my IAX clients (firefly) via the Asterisk dialplan all works great. However trying to initiate a call via the manager interface to the IAX client using the following command results in an error: Action: Originate Channel: IAX2/7000 Extension: 7000 Context: local Priority: 1 ActionID: 1 The error I get in the CLI is Jun 17 08:18:36 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by #IP: No compatible Codecs Does anyone have any ideas. Thanks in advance Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 no compatible codecs
iax2 debug is your friend, looks at the capibilities asterisk is sending in it's NEW message Jason Penton wrote: Hi Adam Done all that but still the same problem. Do you have any other ideas? Cheers Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart Sent: 17 June 2004 08:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 no compatible codecs check under your network settings that you have all the codecs selected and obviously type IAX Jason Penton wrote: Hi All I have a strange problem using IAX2. When placing a call to my IAX clients (firefly) via the Asterisk dialplan all works great. However trying to initiate a call via the manager interface to the IAX client using the following command results in an error: Action: Originate Channel: IAX2/7000 Extension: 7000 Context: local Priority: 1 ActionID: 1 The error I get in the CLI is Jun 17 08:18:36 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by #IP: No compatible Codecs Does anyone have any ideas. Thanks in advance Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pri with TE410P not working (Austria)
hi all, i am trying to get my TE410P (see previous posts) working in Austria (telekom Austria - i am still waiting for an answer for my questions). my /etc/zaptel.conf looks like span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=at defaultzone=at after modprobe wct4xxp and ztcfg -s and ztcfg -v i'll get: TE410P: Disabling interrupts since there are no active spans Registered tone zone 9 (Austria) TE410P: Span 1 configured for CCS/HDB3/CRC4 SPAN 1: Primary Sync Source TE410P: Span 2 configured for CCS/HDB3/CRC4 SPAN 2: Secondary Sync Source ... on the card i can see the two leds pulsing red (i think thats the yellow alaram - or i am wrong) ? then i start asterisk with this zaptel.conf [channels] switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 1 context = default channel = 1-15,17-31 switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 2 context=default channel = 32-46,48-62 on the asterisk prompt i turn on pri intense debugging for span 1 or 2 (both the same result) and i will get (za 2 messages per second): Sending Set Asynchronous Balanced Mode Extended [ 00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data Urgent handler ... and i have no idea what this means can someone please help me to get this working ? best regards Wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Soekris Engineering net4801
John Bittner wrote: Hi, I have it working great. I have debian running on it with music on hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with calls on all 10 phones at the same time through voicepulse with no issues. I ran top with all the phones running and I was only up to 45% cpu. Seems to run ok but I am still in the testing phase. Great... Have you tried to connect a X100P or TDM400P to it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Accepting SIP calls from unregistered gateways
Hi, Is there a way to accept SIP calls from unregistered gateways? autocreatpeer=yes seems to disable checking credentials but the originating gateway is still required to register itself with a username and password (which can be anything since it won't check it). I like to be able to receive the call from any gateway without them having to register even, just like a Cisco gateway that you can terminate a call from clients who are not registered. Is such thing possible with Asterisk? Best regards, Axel
Re: [Asterisk-Users] pri with TE410P not working (Austria)
On Thu, 17 Jun 2004, Wolfgang Pichler wrote: ... on the card i can see the two leds pulsing red (i think thats the yellow alaram - or i am wrong) ? Are you sure it is not a red alarm? That would indicate a loss of link. I think you can check with the command zttool. Are you sure the cables are correct? Have you set the jumpers on the card to E1 and not left them on T1? I think the leds should turn green when the card senses a correct carrier and framing on the lines. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZAPHFC - only for * 0.7.2?
I've got Zaphfc working running Asterisk v. 0.7.2 Then I have tried with Asterisk V. 1.0 and the latest from CVS - with no succes. Has anybody got zaphfc working with newer version than 0.7.2 zaphfc is in bri-stuff from www.junghanns.net --- or in a patched version at http://capi4linux.thepenguin.de/download/asterisk/. I downloaded the latter and let the ./download.sh and ./compile.sh scripts run normally. Then I install zaptel.o and zaphfc.o to /lib/modules/kernelversion/misc and do the usual mambo in /etc/modules to run ztcfg after loading zaphfc and to load zaptel before zaphfc: pre-install zaphfc /sbin/modprobe zaptel post-install zaphfc /sbin/ztcfg -v Now I go to a different directory and do a CVS checkout of Asterisk head. Just before compiling, I replace channels/chan_zap.c with bri-stuff-0.0.2a-pp/asterisk/channels/chan.zap.c. I then change the lines of the form static ast_mutex_t usecnt_lock = AST_MUTEX_INITIALIZER; into AST_MUTEX_DEFINE_STATIC(usecnt_lock); and compile install. And voila, now I have an Asterisk from (almost) CVS HEAD working with zaphfc. The real solution would have been to apply all the patches from bri-stuff*/libpri.patch to libpri in CVS. After looking at how much has been changed and considering that I don't have a clue about q.921 and q.931 I decided to not doing it that way :-) Also, I'd thing it would be better if KaPeJot put's his software into some CVS so that more than one person can add changes and keep things up-to-date. Greetings, Holger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling the firefly network?
Is there a way to register with or call the firefly network from an Asterisk server. It would be pretty cool if you could gateway calls onto it. Have a nice day, -- Martijn van Oosterhout ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 no compatible codecs
Hi Adam Thanks - Here are the two attempts: This is the first one where * dials firefly via the dialplan (which works fine): Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 4 DCall: 0 [146.231.125.65:4569] VERSION : 2 CALLED NUMBER : s CALLING NUMBER : 7001 CALLING NAME: Alfredo+Terzoli LANGUAGE: en FORMAT : 4 CAPABILITY : 2147483647 ADSICPE : 2 DATE TIME : 147935435 Now the following output is when I use the manager ORIGINATE command: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 4 DCall: 0 [146.231.125.65:4569] VERSION : 2 CALLED NUMBER : s LANGUAGE: en FORMAT : 64 CAPABILITY : 2147483647 ADSICPE : 0 DATE TIME : 147935484 Jun 17 10:07:57 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by 146.231.125.65: No compatible Codecs I can see the inconsitency with the FORMAT header of the two setup messages. According to the IAX protocol spec. The FORMAT (0x4) represents G.711 U-LAW, which is exactly what the resulting call uses. However, the funny thing is that the protocol spec has no entry for FORMAT(0x64) in the second message - an undefined format. The quesiton is how the * manager API causes * to inititiate an IAX call with this FORMAT type (0x64)??? An how we can fix it ???. Any ideas, anyone Thanks again Adam for the help Cheers Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart Sent: 17 June 2004 09:19 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 no compatible codecs iax2 debug is your friend, looks at the capibilities asterisk is sending in it's NEW message Jason Penton wrote: Hi Adam Done all that but still the same problem. Do you have any other ideas? Cheers Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart Sent: 17 June 2004 08:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 no compatible codecs check under your network settings that you have all the codecs selected and obviously type IAX Jason Penton wrote: Hi All I have a strange problem using IAX2. When placing a call to my IAX clients (firefly) via the Asterisk dialplan all works great. However trying to initiate a call via the manager interface to the IAX client using the following command results in an error: Action: Originate Channel: IAX2/7000 Extension: 7000 Context: local Priority: 1 ActionID: 1 The error I get in the CLI is Jun 17 08:18:36 WARNING[180236]: chan_iax2.c:4534 socket_read: Call rejected by #IP: No compatible Codecs Does anyone have any ideas. Thanks in advance Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pri with TE410P not working (Austria)
Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43: On Thu, 17 Jun 2004, Wolfgang Pichler wrote: ... on the card i can see the two leds pulsing red (i think thats the yellow alaram - or i am wrong) ? Are you sure it is not a red alarm? That would indicate a loss of link. I think you can check with the command zttool. you are right - its a red alarm - zttool says Red Alarm/Not Open Are you sure the cables are correct? Have you set the jumpers on the card to E1 and not left them on T1? The jumpers are on E1 - the cables should be ok (they are working with other hardware) - and the card is directly connected to a simens ULAF+ STU Desktop (can't really find much information about this device on the net) - which turns off a red led when i load the driver and do a ztcfg. I think the leds should turn green when the card senses a correct carrier and framing on the lines. green is always a wounderful color ;-) so, what else could cause this ? wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] embedded Asterisk
Hi, Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at 233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which is a downstripped Debian ( 64 MB) on a readonly ext2 filesystem, you should be grand. Installing asterisk + some extra stuff will probably require, that you have at least a 128MB or 256MB flash or so. Dont go for stripped down but complete distributions which include a lot of stuff that you dont need, e.g. gcc. Go for a rescue system, like i used the SuSE rescue system (14 mb), then you can add what you need (sshd,...) and compile asterisk on another box and then just copy it. My compressed ramdisk image is 32 mb, including all voice prompts and some mp3s for MOH. There are actually quite some board around on that CPU, like Soekris, pcengines and i think also Mikrotik at prices from 120EUR and up. I just put together the demo system for Linuxtag: - Via EPIA 5000 (C3-533), EUR 80,- - Morex case with external power supply, EUR 80,- - some old 256 mb SDRAMM - 128 MB USB memory stick, EUR 30,- - 1 quadBRI (could also easily handle an octoBRI, or a PRI card, with the dual riser pci card you can use 2 cards) The C3-533 is an i586 CPU. According to show translation it needs 30 ms for transcoding 1 channel from g711 to gsm (and vice versa). So, neglecting any overhead caused by channel handling it could transcode 30 channels to gsm. Linux BIOS has support for the EPIA boards, so you can speed up booting very much and also disable the VGA port (very useful for production deployments). I'm running pebble on a pcengines board, just needed to customize the kernel a bit, haven't been testing asterisk on that yet, but i definatly will in the sooner future. Kind regards, Martin List-Petersen martin (at) list (dash) petersen (dot) net best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pri with TE410P not working (Austria)
What is in your config file ? zaptel.conf ? also, check the crc4 settings and maybe the wire you are using is wrong since some equippment needs crossed wires, some needs straight wires. Crossed would be 1-4 2-5 cheers Michael On Thu, 2004-06-17 at 10:28, Wolfgang Pichler wrote: Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43: On Thu, 17 Jun 2004, Wolfgang Pichler wrote: ... on the card i can see the two leds pulsing red (i think thats the yellow alaram - or i am wrong) ? Are you sure it is not a red alarm? That would indicate a loss of link. I think you can check with the command zttool. you are right - its a red alarm - zttool says Red Alarm/Not Open Are you sure the cables are correct? Have you set the jumpers on the card to E1 and not left them on T1? The jumpers are on E1 - the cables should be ok (they are working with other hardware) - and the card is directly connected to a simens ULAF+ STU Desktop (can't really find much information about this device on the net) - which turns off a red led when i load the driver and do a ztcfg. I think the leds should turn green when the card senses a correct carrier and framing on the lines. green is always a wounderful color ;-) so, what else could cause this ? wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cost of IP Phones, or Isn't It Just Software?
On 16/06/2004 at 22:53 Jay Milk wrote: You're correct -- I believe I pointed out in my original post that there is a $200+ difference between a cordless Cisco with/without software. And that's plain ridiculous. Plus, the phone alone isn't worth $500 in hardware -- so we're obviously dealing with GREED here. My knee-jerk response to such business tactics always has been to do it better and cheaper. Six years ago, I was talking to IT personel in industry X. There were two established mainframe solutions in that industry serving 80% of the market, costing $50K-$75K start-up cost per location, plus $1K+ per seat. Never mind the $10K-$15K monthly maintenance cost. Never mind that everyone had to be able to work a terminal with a lovely amber on black, text-based GUI. snip for brevity I think you're missing the point. When you develop hardware or software you need to recoup the cost of development (the period in which you aren't selling anything, so not making any money). Now Cisco has it's fingers in many pies so they aren't going to suffer to much from that now, but they do have to fund development. Secondly, Cisco don't really care if their phones are out of your price range, they are typically sold as part of a solution costing 10's of 1000's or 100's of 1000's of USD/GBP/EUR and (most probably) with big discounts. Thirdly, If I make a device at a cost of $5 and sell it for $500, some people will buy it, up to the point where someone builds a similar device and sells it for $150 ...You have a choice. companies are not charities, they do this to make money. This is what we call capitalism. I don't want to dig at your business, and this isn't intended to but.. what you did is look at what was already on offer and it's costs, how it worked etc and built a cheaper solution. The reason you could do this is because you had the exposure to the 'system' as was.. i.e. You looked at it and said 'I can do that cheaper' but without that original system you probably wouldn't have. One final point... There are some companies that have this weird feeling that anything under a certain amount must be cheap and nasty and not work properly. These people are fools imho, but they do exist...and they wont buy an cheap phone, they'll buy an expensive phone, regardless of it's ability... as we've seen recently some governments will even buy helicopters that can't fly in fog or where it's sandy for silly money... Now I feel dirty... Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pri with TE410P not working (Austria)
On Thu, 17 Jun 2004, Wolfgang Pichler wrote: Are you sure the cables are correct? Have you set the jumpers on the card to E1 and not left them on T1? The jumpers are on E1 - the cables should be ok (they are working with other hardware) - and the card is directly connected to a simens ULAF+ STU Desktop (can't really find much information about this device on the net) - which turns off a red led when i load the driver and do a ztcfg. Then the tx (from TE410P to the Siemens equipment) circuit is ok but the rx may not be. I think the leds should turn green when the card senses a correct carrier and framing on the lines. green is always a wounderful color ;-) so, what else could cause this ? I'd try to find out if the cable is wired the way the TE410P expects it to be. Do you know the pinout of both ends of the cables? RX (from the TE410P point of view) should be on the pins 1-2 at the TE410P end and TX on 4-5. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LDAP synchronization script
Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Thanks, -D ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pri with TE410P not working (Austria)
Am Do, den 17.06.2004 schrieb Michael Bielicki um 10:32: What is in your config file ? i've already posted my config in my first post - but here is my /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=at defaultzone=at and my /etc/asterisk/zaptel.conf [channels] switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 1 context = default channel = 1-15,17-31 switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 2 context=default channel = 32-46,48-62 zaptel.conf ? also, check the crc4 settings which crc4 settings ? - i've thought crc4 could only be turned on or off ? and maybe the wire you are using is wrong since some equippment needs crossed wires, some needs straight wires. Crossed would be 1-4 2-5 i've already tried to use crossed wires - didn't worked either cheers Michael On Thu, 2004-06-17 at 10:28, Wolfgang Pichler wrote: Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43: On Thu, 17 Jun 2004, Wolfgang Pichler wrote: ... on the card i can see the two leds pulsing red (i think thats the yellow alaram - or i am wrong) ? Are you sure it is not a red alarm? That would indicate a loss of link. I think you can check with the command zttool. you are right - its a red alarm - zttool says Red Alarm/Not Open Are you sure the cables are correct? Have you set the jumpers on the card to E1 and not left them on T1? The jumpers are on E1 - the cables should be ok (they are working with other hardware) - and the card is directly connected to a simens ULAF+ STU Desktop (can't really find much information about this device on the net) - which turns off a red led when i load the driver and do a ztcfg. I think the leds should turn green when the card senses a correct carrier and framing on the lines. green is always a wounderful color ;-) so, what else could cause this ? wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling the firefly network?
Martijn van Oosterhout wrote: Is there a way to register with or call the firefly network from an Asterisk server. It would be pretty cool if you could gateway calls onto it. Have a nice day, You can register and dial out with * like on other IAX services. You can verify it by changing the network settings from Firefly to IAX Firefly's network tab. On * the connection gets lost if someone sends an IM via Firefly client. I 've added speex and iLBC to the allowed codecs in iax.conf. I can call and receive to and from freshtel numbers, didn't check PSTN gateway yet. jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LDAP synchronization script
I think there is a odbc driver for ldap. at least I remember that I saw one a while ago. You could combine that with ast_data and off you fly Just my 2 cents (EUR) Michael On Thu, 2004-06-17 at 10:41, David Hajek wrote: Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Thanks, -D ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pri with TE410P not working (Austria)
Throw out the yellow. Check if for sure the other side is using crc4. On Thu, 2004-06-17 at 10:45, Wolfgang Pichler wrote: Am Do, den 17.06.2004 schrieb Michael Bielicki um 10:32: What is in your config file ? i've already posted my config in my first post - but here is my /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=at defaultzone=at and my /etc/asterisk/zaptel.conf [channels] switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 1 context = default channel = 1-15,17-31 switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 2 context=default channel = 32-46,48-62 zaptel.conf ? also, check the crc4 settings which crc4 settings ? - i've thought crc4 could only be turned on or off ? and maybe the wire you are using is wrong since some equippment needs crossed wires, some needs straight wires. Crossed would be 1-4 2-5 i've already tried to use crossed wires - didn't worked either cheers Michael On Thu, 2004-06-17 at 10:28, Wolfgang Pichler wrote: Am Do, den 17.06.2004 schrieb Peter Svensson um 9:43: On Thu, 17 Jun 2004, Wolfgang Pichler wrote: ... on the card i can see the two leds pulsing red (i think thats the yellow alaram - or i am wrong) ? Are you sure it is not a red alarm? That would indicate a loss of link. I think you can check with the command zttool. you are right - its a red alarm - zttool says Red Alarm/Not Open Are you sure the cables are correct? Have you set the jumpers on the card to E1 and not left them on T1? The jumpers are on E1 - the cables should be ok (they are working with other hardware) - and the card is directly connected to a simens ULAF+ STU Desktop (can't really find much information about this device on the net) - which turns off a red led when i load the driver and do a ztcfg. I think the leds should turn green when the card senses a correct carrier and framing on the lines. green is always a wounderful color ;-) so, what else could cause this ? wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pri with TE410P not working (Austria)
Am Do, den 17.06.2004 schrieb Peter Svensson um 10:38: On Thu, 17 Jun 2004, Wolfgang Pichler wrote: Are you sure the cables are correct? Have you set the jumpers on the card to E1 and not left them on T1? The jumpers are on E1 - the cables should be ok (they are working with other hardware) - and the card is directly connected to a simens ULAF+ STU Desktop (can't really find much information about this device on the net) - which turns off a red led when i load the driver and do a ztcfg. Then the tx (from TE410P to the Siemens equipment) circuit is ok but the rx may not be. but the same cable works great with an other hardware (a Parlay i60) I think the leds should turn green when the card senses a correct carrier and framing on the lines. green is always a wounderful color ;-) so, what else could cause this ? I'd try to find out if the cable is wired the way the TE410P expects it to be. Do you know the pinout of both ends of the cables? RX (from the TE410P point of view) should be on the pins 1-2 at the TE410P end and TX on 4-5. would be a great thing if you can find something more than i found best regards wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail
HI ALL; Is asterisk voicemail service can be run under H323 or it just run under SIP. mohammad
RE: [Asterisk-Users] oh323
Can I just pay you to fix it for me. I cant see anywhere where I use the debug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 11:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Have you enabled non-default compiler flags in Asterisk's top-level Makefile (e.g DEBUG_THREAD)? Michael. Michael M. Saunders wrote: Have recompiled a few times any ideas? *CLI oh323 debug toggle Verbose debug info for OpenH323 channel turned on. *CLI Jun 17 23:28:55 ERROR[20499]: chan_oh323.c:2297 ast_oh323_new: Internal channel initialization failed. Bad binary? *CLI set verbose 4 *CLI Jun 17 23:29:24 ERROR[21523]: chan_oh323.c:2297 ast_oh323_new: Internal channel initialization failed. Bad binary? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 6:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Update to version 0.6.2a. It compiles with today's Asterisk CVS HEAD. http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asteris k-oh323-0.6.2a.tar.gz Michael. Michael M. Saunders wrote: The other problem is that version. Doesn't seem to work well with e1'. I rephrase it changes everything back to t1. Is there any way I can get it working with the latest version of cvs head -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael M. Saunders Sent: Wednesday, 16 June 2004 7:55 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] oh323 Thanks that worked. Is there any plans to make it work with the lastest cvs head -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 1:10 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 cvs co -D 2004-06-07 asterisk Michael M. Saunders wrote: I grabbed the lastest CVS and it stilled failed. Would you be able to give me the command to get 2004-06-07 Because when I login I can only get it by release numbers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Tuesday, 15 June 2004 9:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 You are trying to compile it with an out-dated Asterisk source tree. Use Asterisk CVS HEAD checkout of 2004-06-07. Michael. Michael M. Saunders wrote: Does anyone have any ideas why this is failing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael M. Saunders Sent: Monday, 14 June 2004 6:30 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] oh323 debian:/usr/src/asterisk-oh323-0.6.2# make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1385: too few arguments to function `ast_queue_control' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1417: too few arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_read': chan_oh323.c:1855: too few arguments to function `ast_dsp_process' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 debian:/usr/src/asterisk-oh323-0.6.2# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Grimshaw Sent: Monday, 14 June 2004 6:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 On Mon, 14 Jun 2004 18:06:25 +1000, Michael M. Saunders [EMAIL PROTECTED] wrote: This module wont compile can anyone give me any assistance Sure, what error messages is it giving you Michael? -- ./M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pri with TE410P not working (Austria)
On Thu, 17 Jun 2004, Wolfgang Pichler wrote: but the same cable works great with an other hardware (a Parlay i60) You could try a loopback plug to make sure your TE410P does not have a damaged receiver. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failed to authenticate on INVITE
At 16:49 16/06/2004 -0400, Eric wrote: I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error Failed to authenticate on INVITE trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. include the line in sip.conf for each user the call insecure=yes ; To match a peer based by IP address only and not peer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with PRI with T410 messages
Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on and the phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Phone Services TERRACOM Broadband [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 10:56 AM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: Soekris Engineering net4801 (Senad Jordanovic) 2. Accepting SIP calls from unregistered gateways (Axel) 3. Re: pri with TE410P not working (Austria) (Peter Svensson) 4. Re: ZAPHFC - only for * 0.7.2? (Holger Schurig) 5. Calling the firefly network? (Martijn van Oosterhout) 6. RE: IAX2 no compatible codecs (Jason Penton) 7. Re: pri with TE410P not working (Austria) (Wolfgang Pichler) 8. Re: embedded Asterisk (Klaus-Peter Junghanns) 9. Re: pri with TE410P not working (Austria) (Michael Bielicki) 10. RE: Cost of IP Phones, or Isn't It Just Software? (Andy Powell) 11. Re: pri with TE410P not working (Austria) (Peter Svensson) --__--__-- Message: 1 From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Soekris Engineering net4801 Date: Thu, 17 Jun 2004 08:34:01 +0100 Reply-To: [EMAIL PROTECTED] John Bittner wrote: Hi, I have it working great. I have debian running on it with music on hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with calls on all 10 phones at the same time through voicepulse with no issues. I ran top with all the phones running and I was only up to 45% cpu. Seems to run ok but I am still in the testing phase. Great... Have you tried to connect a X100P or TDM400P to it? --__--__-- Message: 2 From: Axel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 17 Jun 2004 03:43:12 -0400 Subject: [Asterisk-Users] Accepting SIP calls from unregistered gateways Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable Hi, Is there a way to accept SIP calls from unregistered gateways? autocreatpeer=3Dyes seems to disable checking credentials but the = originating gateway is still required to register itself with a username = and password (which can be anything since it won't check it). I like to be able to receive the call from any gateway without them = having to register even, just like a Cisco gateway that you can = terminate a call from clients who are not registered. Is such thing = possible with Asterisk? Best regards, Axel --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 4.0 Transitional//EN HTMLHEAD META http-equiv=3DContent-Type content=3Dtext/html; = charset=3Diso-8859-1 META content=3DMSHTML 6.00.2800.1400 name=3DGENERATOR STYLE/STYLE /HEAD BODY bgColor=3D#ff DIVFONT face=3DArial size=3D2Hi,/FONT/DIV DIVFONT face=3DArial size=3D2Is there a way to accept SIP calls from = unregistered gateways?/FONT/DIV DIVFONT face=3DArial size=3D2autocreatpeer=3Dyes seems to disable = checking=20 credentials but the originating gateway is still required to register = itself=20 with a username and password (which can be anything since it won't check = it)./FONT/DIV DIVFONT face=3DArial size=3D2I like to be able to receive the call = from any=20 gateway without them having to register even, just like a Cisco gateway = that you=20 can terminate a call from clients who are not registered.nbsp; Is such = thing=20 possible with Asterisk?/FONT/DIV DIVFONT face=3DArial size=3D2/FONTnbsp;/DIV DIVFONT face=3DArial size=3D2Best regards,/FONT/DIV DIVnbsp;/DIV DIVFONT face=3DArial size=3D2AxelBR/FONT/DIV/BODY/HTML --=_NextPart_000_0351_01C4541D.36B45830-- --__--__-- Message: 3 Date: Thu, 17 Jun 2004
Re: [Asterisk-Users] oh323
Did you compile the channel driver with the sources of the running Asterisk? This is happening because of a mismatch between the include Asterisk files used to compile asterisk-oh323 and the running Asterisk. Make sure that you have removed any previous version of Asterisk (including header files and modules) before trying to install a fresh copy of it or compile a new asterisk-oh323. Michael. Michael M. Saunders wrote: Can I just pay you to fix it for me. I cant see anywhere where I use the debug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 11:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Have you enabled non-default compiler flags in Asterisk's top-level Makefile (e.g DEBUG_THREAD)? Michael. Michael M. Saunders wrote: Have recompiled a few times any ideas? *CLI oh323 debug toggle Verbose debug info for OpenH323 channel turned on. *CLI Jun 17 23:28:55 ERROR[20499]: chan_oh323.c:2297 ast_oh323_new: Internal channel initialization failed. Bad binary? *CLI set verbose 4 *CLI Jun 17 23:29:24 ERROR[21523]: chan_oh323.c:2297 ast_oh323_new: Internal channel initialization failed. Bad binary? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 6:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Update to version 0.6.2a. It compiles with today's Asterisk CVS HEAD. http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asteris k-oh323-0.6.2a.tar.gz Michael. Michael M. Saunders wrote: The other problem is that version. Doesn't seem to work well with e1'. I rephrase it changes everything back to t1. Is there any way I can get it working with the latest version of cvs head -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael M. Saunders Sent: Wednesday, 16 June 2004 7:55 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] oh323 Thanks that worked. Is there any plans to make it work with the lastest cvs head -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 1:10 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 cvs co -D 2004-06-07 asterisk Michael M. Saunders wrote: I grabbed the lastest CVS and it stilled failed. Would you be able to give me the command to get 2004-06-07 Because when I login I can only get it by release numbers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Tuesday, 15 June 2004 9:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 You are trying to compile it with an out-dated Asterisk source tree. Use Asterisk CVS HEAD checkout of 2004-06-07. Michael. Michael M. Saunders wrote: Does anyone have any ideas why this is failing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael M. Saunders Sent: Monday, 14 June 2004 6:30 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] oh323 debian:/usr/src/asterisk-oh323-0.6.2# make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1385: too few arguments to function `ast_queue_control' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1417: too few arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_read': chan_oh323.c:1855: too few arguments to function `ast_dsp_process' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 debian:/usr/src/asterisk-oh323-0.6.2# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Grimshaw Sent: Monday, 14 June 2004 6:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 On Mon, 14 Jun 2004 18:06:25 +1000, Michael M. Saunders [EMAIL PROTECTED] wrote: This module wont compile can anyone give me any assistance Sure, what error messages is it giving you Michael? -- ./M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] Outgoing DTMF when using BRI i4l (Eicon Diva) - problems
On Tue, 2004-06-15 at 17:44, Mark Elkins wrote: On Tue, 2004-06-15 at 11:43, Shaun Ewing wrote: This is an issues with DTMF clamping, you need to use chan_capi to get DTMF working correctly. That's the last thing I wanted to hear :-( The jist of this is that i4l does not allow outgoing DTMF ??? ie - its broken??? Has anyone got the combination of Grandstream (I think this is irrelevent) + ISDN BRI (Dumb Cards - all seem to cause the problem) + i4l + outgoing DTMF working at all? What Version of Asterisk? So far - people who I've asked say No -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UIP200
The disconnect issue also still exists (for me) with 4.55 firmware. I can use the uniden to call another local sip phone (with canreinvite=no), and leave both phones off the hook for as long as I like. However, if I use the Uniden to call an PSTN number (via tdm400p fxo), then the uniden will start playing a congestion tone (not from asterisk), and cease tx/rx audio. After this happens, a 'show channels' will show that the call is still active. Make sure you have VAD turned off and silence suppression turned off * may not be getting a continuous RTP stream Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] oh323
What is the easiest way to guarantee everything is gone rm -f /usr/lib/asterisk is there anything else -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Thursday, 17 June 2004 7:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Did you compile the channel driver with the sources of the running Asterisk? This is happening because of a mismatch between the include Asterisk files used to compile asterisk-oh323 and the running Asterisk. Make sure that you have removed any previous version of Asterisk (including header files and modules) before trying to install a fresh copy of it or compile a new asterisk-oh323. Michael. Michael M. Saunders wrote: Can I just pay you to fix it for me. I cant see anywhere where I use the debug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 11:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Have you enabled non-default compiler flags in Asterisk's top-level Makefile (e.g DEBUG_THREAD)? Michael. Michael M. Saunders wrote: Have recompiled a few times any ideas? *CLI oh323 debug toggle Verbose debug info for OpenH323 channel turned on. *CLI Jun 17 23:28:55 ERROR[20499]: chan_oh323.c:2297 ast_oh323_new: Internal channel initialization failed. Bad binary? *CLI set verbose 4 *CLI Jun 17 23:29:24 ERROR[21523]: chan_oh323.c:2297 ast_oh323_new: Internal channel initialization failed. Bad binary? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 6:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Update to version 0.6.2a. It compiles with today's Asterisk CVS HEAD. http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asteris k-oh323-0.6.2a.tar.gz Michael. Michael M. Saunders wrote: The other problem is that version. Doesn't seem to work well with e1'. I rephrase it changes everything back to t1. Is there any way I can get it working with the latest version of cvs head -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael M. Saunders Sent: Wednesday, 16 June 2004 7:55 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] oh323 Thanks that worked. Is there any plans to make it work with the lastest cvs head -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 1:10 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 cvs co -D 2004-06-07 asterisk Michael M. Saunders wrote: I grabbed the lastest CVS and it stilled failed. Would you be able to give me the command to get 2004-06-07 Because when I login I can only get it by release numbers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Tuesday, 15 June 2004 9:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 You are trying to compile it with an out-dated Asterisk source tree. Use Asterisk CVS HEAD checkout of 2004-06-07. Michael. Michael M. Saunders wrote: Does anyone have any ideas why this is failing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael M. Saunders Sent: Monday, 14 June 2004 6:30 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] oh323 debian:/usr/src/asterisk-oh323-0.6.2# make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1385: too few arguments to function `ast_queue_control' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1417: too few arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_read': chan_oh323.c:1855: too few arguments to function `ast_dsp_process' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 debian:/usr/src/asterisk-oh323-0.6.2# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Grimshaw Sent: Monday, 14 June 2004 6:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 On Mon, 14 Jun 2004 18:06:25 +1000, Michael M.
Re: [Asterisk-Users] embedded Asterisk
Hi! I will use it as simple ivr ...get the call from fxo gateway port ..give some options and rings the recepcionist phone. I have a x100p here and the thin client have a pci slot...maybe i can use it...maybe...not...i will test The main reason is to free a p4 2.0 ..that is runing * now... i think that it is to much only to say hello...press 1. :-) Miklos - Original Message - From: Stefan de Konink [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 16, 2004 8:22 PM Subject: Re: [Asterisk-Users] embedded Asterisk Probably the best thing to do is to build a uClibc tree, disable some Asterisk codecs (which don't want to compile, first run) compile again and run. Tomorrow I'm going to do the samething for an Epia-MII 1,2GHz/512MB/512MB-CF. Another tip :P Don't compile on flash... just make a tree on your harddisk. And copy the required binaries and libs to a root tree and attach a kernel. Look at some different Filesystems too, depending on for needs Ext2/Minix/CramFS. Btw. for what purpose do you want to run the box? I can imagine that a few voicemail messages can float the system. And if SIP is only required you should probably use SER for the project. I want to try out the VOCAL footprint too but didn't had the time to do that yet. Stefan listas iPfone wrote: Hi All, I have a thin cliente here that i want to run asterisk: - National Semicondudor Geode GX1 266MHz Geode 266MHz single chip - NS Cx5530a Southbridge National Semiconductors SC2200 - NS PC97317 in chipset - 32MB Compact Flash - 64MB Ram - 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816 Some tip? I have a ideflash adaptor to make the install... I need recomendations in Linux distro... asterisk min. install ...etc..any info i can get. Thanks for any help Miklos Atenciosamente Cláudio Miklos * iP FONE *Telefonia IP Rua Caio Graco 735 São Paulo SP ( BR - 55 11 3801-3702 ( USA - 1 360-968-1591 ( FWD - 64662 ( sip:[EMAIL PROTECTED] www.ipfone.com.br http://www.ipfone.com.br [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LDAP synchronization script
David Hajek Sent: Thursday, June 17, 2004 2:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] LDAP synchronization script Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Not aware of any scripts like that, but... you could use the odbc support in asterisk in conjunction with some slick odbc-ldap connectivity. Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323
And rm -rf /usr/include/asterisk Michael M. Saunders wrote: What is the easiest way to guarantee everything is gone rm -f /usr/lib/asterisk is there anything else -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Thursday, 17 June 2004 7:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Did you compile the channel driver with the sources of the running Asterisk? This is happening because of a mismatch between the include Asterisk files used to compile asterisk-oh323 and the running Asterisk. Make sure that you have removed any previous version of Asterisk (including header files and modules) before trying to install a fresh copy of it or compile a new asterisk-oh323. Michael. Michael M. Saunders wrote: Can I just pay you to fix it for me. I cant see anywhere where I use the debug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 11:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Have you enabled non-default compiler flags in Asterisk's top-level Makefile (e.g DEBUG_THREAD)? Michael. Michael M. Saunders wrote: Have recompiled a few times any ideas? *CLI oh323 debug toggle Verbose debug info for OpenH323 channel turned on. *CLI Jun 17 23:28:55 ERROR[20499]: chan_oh323.c:2297 ast_oh323_new: Internal channel initialization failed. Bad binary? *CLI set verbose 4 *CLI Jun 17 23:29:24 ERROR[21523]: chan_oh323.c:2297 ast_oh323_new: Internal channel initialization failed. Bad binary? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 6:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Update to version 0.6.2a. It compiles with today's Asterisk CVS HEAD. http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asteris k-oh323-0.6.2a.tar.gz Michael. Michael M. Saunders wrote: The other problem is that version. Doesn't seem to work well with e1'. I rephrase it changes everything back to t1. Is there any way I can get it working with the latest version of cvs head -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael M. Saunders Sent: Wednesday, 16 June 2004 7:55 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] oh323 Thanks that worked. Is there any plans to make it work with the lastest cvs head -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Wednesday, 16 June 2004 1:10 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 cvs co -D 2004-06-07 asterisk Michael M. Saunders wrote: I grabbed the lastest CVS and it stilled failed. Would you be able to give me the command to get 2004-06-07 Because when I login I can only get it by release numbers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Tuesday, 15 June 2004 9:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 You are trying to compile it with an out-dated Asterisk source tree. Use Asterisk CVS HEAD checkout of 2004-06-07. Michael. Michael M. Saunders wrote: Does anyone have any ideas why this is failing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael M. Saunders Sent: Monday, 14 June 2004 6:30 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] oh323 debian:/usr/src/asterisk-oh323-0.6.2# make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1385: too few arguments to function `ast_queue_control' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1417: too few arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_read': chan_oh323.c:1855: too few arguments to function `ast_dsp_process' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 debian:/usr/src/asterisk-oh323-0.6.2# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Grimshaw Sent: Monday, 14 June 2004 6:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 On Mon, 14 Jun 2004 18:06:25 +1000, Michael M. Saunders
[Asterisk-Users] HFC ISDN card with bristuff from junghanns.net?
Hi, has anyone succesfully installed such scenario ? I'm having problem with Award bios mb pc's... it do works with others, what's your idea ? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with PRI with T410 messages
This is a problem I pointed out to Digium a while back, but I am not sure Markster understood the issue and I didn't really have the time to follow it up. It does need fixing though, as it is a major drawback in the current architecture. Rgds Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aimable Sent: 17 June 2004 10:29 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problems with PRI with T410 messages Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on and the phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Phone Services TERRACOM Broadband [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 10:56 AM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: Soekris Engineering net4801 (Senad Jordanovic) 2. Accepting SIP calls from unregistered gateways (Axel) 3. Re: pri with TE410P not working (Austria) (Peter Svensson) 4. Re: ZAPHFC - only for * 0.7.2? (Holger Schurig) 5. Calling the firefly network? (Martijn van Oosterhout) 6. RE: IAX2 no compatible codecs (Jason Penton) 7. Re: pri with TE410P not working (Austria) (Wolfgang Pichler) 8. Re: embedded Asterisk (Klaus-Peter Junghanns) 9. Re: pri with TE410P not working (Austria) (Michael Bielicki) 10. RE: Cost of IP Phones, or Isn't It Just Software? (Andy Powell) 11. Re: pri with TE410P not working (Austria) (Peter Svensson) --__--__-- Message: 1 From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Soekris Engineering net4801 Date: Thu, 17 Jun 2004 08:34:01 +0100 Reply-To: [EMAIL PROTECTED] John Bittner wrote: Hi, I have it working great. I have debian running on it with music on hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with calls on all 10 phones at the same time through voicepulse with no issues. I ran top with all the phones running and I was only up to 45% cpu. Seems to run ok but I am still in the testing phase. Great... Have you tried to connect a X100P or TDM400P to it? --__--__-- Message: 2 From: Axel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 17 Jun 2004 03:43:12 -0400 Subject: [Asterisk-Users] Accepting SIP calls from unregistered gateways Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable Hi, Is there a way to accept SIP calls from unregistered gateways? autocreatpeer=3Dyes seems to disable checking credentials but the = originating gateway is still required to register itself with a username = and password (which can be anything since it won't check it). I like to be able to receive the call from any gateway without them = having to register even, just like a Cisco gateway that you can = terminate a call from clients who are not registered. Is such thing = possible with Asterisk? Best regards, Axel --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 4.0 Transitional//EN HTMLHEAD META http-equiv=3DContent-Type content=3Dtext/html; = charset=3Diso-8859-1 META content=3DMSHTML 6.00.2800.1400 name=3DGENERATOR STYLE/STYLE /HEAD BODY bgColor=3D#ff DIVFONT face=3DArial size=3D2Hi,/FONT/DIV DIVFONT face=3DArial size=3D2Is there a way to accept SIP calls from = unregistered gateways?/FONT/DIV DIVFONT face=3DArial size=3D2autocreatpeer=3Dyes seems to disable = checking=20 credentials but the originating gateway is still required to register = itself=20 with a username and password (which can be anything since it won't check = it)./FONT/DIV DIVFONT face=3DArial size=3D2I like to be able to
RE: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.n et?
Please can you explain in more details as to what your problem is? I have 2 cards working in one PC, but have had problems with Dell motherboards. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alessio Focardi Sent: 17 June 2004 11:41 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.net? Hi, has anyone succesfully installed such scenario ? I'm having problem with Award bios mb pc's... it do works with others, what's your idea ? Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] embedded Asterisk
google for it :) http://lists.digium.com/pipermail/asterisk-dev/2003-November/002299.html On Jun 17, 2004, at 1:05 AM, listas iPfone wrote: Hi All, I have a thin cliente here that i want to run asterisk: - National Semicondudor Geode GX1 266MHz Geode 266MHz single chip - NS Cx5530a Southbridge National Semiconductors SC2200 - NS PC97317 in chipset - 32MB Compact Flash - 64MB Ram - 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816 Some tip? I have a ideflash adaptor to make the install... I need recomendations in Linux distro... asterisk min. install ...etc..any info i can get. Thanks for any help Miklos Atenciosamente Cláudio Miklos iPFONE Telefonia IP Rua Caio Graco 735 São Paulo SP ( BR - 55 11 3801-3702 ( USA - 1 360-968-1591 ( FWD - 64662 ( sip:[EMAIL PROTECTED] www.ipfone.com.br [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outgoing DTMF when using BRI i4l (Eicon Diva) - problems
On Thu, 2004-06-17 at 04:40, Mark Elkins wrote: On Tue, 2004-06-15 at 17:44, Mark Elkins wrote: On Tue, 2004-06-15 at 11:43, Shaun Ewing wrote: This is an issues with DTMF clamping, you need to use chan_capi to get DTMF working correctly. That's the last thing I wanted to hear :-( The jist of this is that i4l does not allow outgoing DTMF ??? ie - its broken??? Has anyone got the combination of Grandstream (I think this is irrelevent) + ISDN BRI (Dumb Cards - all seem to cause the problem) + i4l + outgoing DTMF working at all? What Version of Asterisk? So far - people who I've asked say No http://www.google.com/search?hl=enlr=ie=UTF-8q=site%3Alists.digium.com+i4l+dtmf+patchbtnG=Search -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone have experience with chan-capi in Australia?
I'm planning a system, just need to know if it works with Telstra's network. Cheers, Clint Sydney, Australia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with PRI with T410 messages
Send traces. - Original Message - From: Aimable [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 17, 2004 6:28 AM Subject: [Asterisk-Users] Problems with PRI with T410 messages Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on and the phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Phone Services TERRACOM Broadband [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 10:56 AM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: Soekris Engineering net4801 (Senad Jordanovic) 2. Accepting SIP calls from unregistered gateways (Axel) 3. Re: pri with TE410P not working (Austria) (Peter Svensson) 4. Re: ZAPHFC - only for * 0.7.2? (Holger Schurig) 5. Calling the firefly network? (Martijn van Oosterhout) 6. RE: IAX2 no compatible codecs (Jason Penton) 7. Re: pri with TE410P not working (Austria) (Wolfgang Pichler) 8. Re: embedded Asterisk (Klaus-Peter Junghanns) 9. Re: pri with TE410P not working (Austria) (Michael Bielicki) 10. RE: Cost of IP Phones, or Isn't It Just Software? (Andy Powell) 11. Re: pri with TE410P not working (Austria) (Peter Svensson) --__--__-- Message: 1 From: Senad Jordanovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Soekris Engineering net4801 Date: Thu, 17 Jun 2004 08:34:01 +0100 Reply-To: [EMAIL PROTECTED] John Bittner wrote: Hi, I have it working great. I have debian running on it with music on hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with calls on all 10 phones at the same time through voicepulse with no issues. I ran top with all the phones running and I was only up to 45% cpu. Seems to run ok but I am still in the testing phase. Great... Have you tried to connect a X100P or TDM400P to it? --__--__-- Message: 2 From: Axel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 17 Jun 2004 03:43:12 -0400 Subject: [Asterisk-Users] Accepting SIP calls from unregistered gateways Reply-To: [EMAIL PROTECTED] This is a multi-part message in MIME format. --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable Hi, Is there a way to accept SIP calls from unregistered gateways? autocreatpeer=3Dyes seems to disable checking credentials but the = originating gateway is still required to register itself with a username = and password (which can be anything since it won't check it). I like to be able to receive the call from any gateway without them = having to register even, just like a Cisco gateway that you can = terminate a call from clients who are not registered. Is such thing = possible with Asterisk? Best regards, Axel --=_NextPart_000_0351_01C4541D.36B45830 Content-Type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: quoted-printable !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 4.0 Transitional//EN HTMLHEAD META http-equiv=3DContent-Type content=3Dtext/html; = charset=3Diso-8859-1 META content=3DMSHTML 6.00.2800.1400 name=3DGENERATOR STYLE/STYLE /HEAD BODY bgColor=3D#ff DIVFONT face=3DArial size=3D2Hi,/FONT/DIV DIVFONT face=3DArial size=3D2Is there a way to accept SIP calls from = unregistered gateways?/FONT/DIV DIVFONT face=3DArial size=3D2autocreatpeer=3Dyes seems to disable = checking=20 credentials but the originating gateway is still required to register = itself=20 with a username and password (which can be anything since it won't check = it)./FONT/DIV DIVFONT face=3DArial size=3D2I like to be able to receive the call = from any=20 gateway without them having to register even, just like a Cisco gateway = that you=20 can terminate a call from clients who are not registered.nbsp; Is such = thing=20 possible with Asterisk?/FONT/DIV
[Asterisk-Users] Cheap (US$120 or less) SIP Phones
These are the three cheap SIP phones that I've used. Grandstream BT10x $65/street Number only LCD Zultys ZIP 2 $100/retail No LCD Uniden UIP 200 $120/retail PoE, built-in switch -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.n et?
Hello Robinson, Thursday, June 17, 2004, 12:42:21 PM, you wrote: RTW Please can you explain in more details as to what your RTW problem is? I have 2 cards working in one PC, but have had RTW problems with Dell motherboards. voice is out of sync, it syncs for some second if I run something over another console, like, for instance a find / then slips away again. I suspect an Irq problem, what do you think ? What kind of problems have you found with dell's ? Tnx for the help ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LDAP synchronization script
I'm planning to incorporate this (native and dynamic) LDAP for my own system on short term. Do you have any LDAP design in mind? Stefan On Thu, 17 Jun 2004, Jeremy Jones wrote: David Hajek Sent: Thursday, June 17, 2004 2:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] LDAP synchronization script Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Not aware of any scripts like that, but... you could use the odbc support in asterisk in conjunction with some slick odbc-ldap connectivity. Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with PRI with T410 messages
This is a problem I pointed out to Digium a while back, but I am not sure Markster understood the issue and I didn't really have the time to follow it up. It does need fixing though, as it is a major drawback in the current architecture. Rgds Tim Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on and the phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Call Proceeding can be sent only by transit network, not by the local switch or pbx. AFAIK, * behavior for this scenario is like as local switch. Certainly, this is not a normal behavior. Regards, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from jung hanns.n et?
Hi Alessio Yes, the problems you report do seem similar to the issues I had. I found on the Dells that the audio prompts were very choppy and played slower than normal. Occasionally there would be 'bursts' oav a second or so of 'good' audio. I also suspected IRQ issues but the Dell Mobos had no way of adjusting them. Best thing is to try and get the card on its own unshared IRQ. If this fails, you either have to try a different pc, or collect 600 euros together and send them to Junghanns.net, and they will send you a quadBRI card that does not have this problem. Rgds Tim -Original Message- From: Alessio Focardi [mailto:[EMAIL PROTECTED] Sent: 17 June 2004 12:19 To: Robinson Tim-W10277; [EMAIL PROTECTED] Subject: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.n et? Hello Robinson, Thursday, June 17, 2004, 12:42:21 PM, you wrote: RTW Please can you explain in more details as to what your problem is? RTW I have 2 cards working in one PC, but have had problems with Dell RTW motherboards. voice is out of sync, it syncs for some second if I run something over another console, like, for instance a find / then slips away again. I suspect an Irq problem, what do you think ? What kind of problems have you found with dell's ? Tnx for the help ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LDAP synchronization script
I think I'll use something from this article - http://www.marko.net/asterisk/archives/0205/0006.html -David -Original Message- From: Stefan de Konink [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 1:12 PM To: David Hajek Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LDAP synchronization script I'm planning to incorporate this (native and dynamic) LDAP for my own system on short term. Do you have any LDAP design in mind? Stefan On Thu, 17 Jun 2004, Jeremy Jones wrote: David Hajek Sent: Thursday, June 17, 2004 2:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] LDAP synchronization script Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Not aware of any scripts like that, but... you could use the odbc support in asterisk in conjunction with some slick odbc-ldap connectivity. Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zapata.conf Signaling for Bulgaria (PSTN: Siemens PABX)
Hi, How to configure our ZAPATA.CONF in case that the PSTN in Bulgaria is based on Siemens equipment? Now my configuration is: [channels] language=en busydetect=no when is yes I have problems with answering of FXO when FXS line is open callprogress=no when is yes I have problems with answering of FXO when FXS line is open ; interfaces for internal analog phones signalling=fxo_ks threewaycalling=yes ; interfaces for external PSTN line signalling=fxs_ks Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] embedded Asterisk
Hi That rescue disk sugestion seems to be very good... Let´s see if i undestood: 1. burn the rescue iso 1. copy the rescue disk to a hard drive 2. compile asterisk 3. copy all to the flash disk It is that simple? Miklos - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 17, 2004 5:11 AM Subject: Re: [Asterisk-Users] embedded Asterisk Hi, Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at 233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which is a downstripped Debian ( 64 MB) on a readonly ext2 filesystem, you should be grand. Installing asterisk + some extra stuff will probably require, that you have at least a 128MB or 256MB flash or so. Dont go for stripped down but complete distributions which include a lot of stuff that you dont need, e.g. gcc. Go for a rescue system, like i used the SuSE rescue system (14 mb), then you can add what you need (sshd,...) and compile asterisk on another box and then just copy it. My compressed ramdisk image is 32 mb, including all voice prompts and some mp3s for MOH. There are actually quite some board around on that CPU, like Soekris, pcengines and i think also Mikrotik at prices from 120EUR and up. I just put together the demo system for Linuxtag: - Via EPIA 5000 (C3-533), EUR 80,- - Morex case with external power supply, EUR 80,- - some old 256 mb SDRAMM - 128 MB USB memory stick, EUR 30,- - 1 quadBRI (could also easily handle an octoBRI, or a PRI card, with the dual riser pci card you can use 2 cards) The C3-533 is an i586 CPU. According to show translation it needs 30 ms for transcoding 1 channel from g711 to gsm (and vice versa). So, neglecting any overhead caused by channel handling it could transcode 30 channels to gsm. Linux BIOS has support for the EPIA boards, so you can speed up booting very much and also disable the VGA port (very useful for production deployments). I'm running pebble on a pcengines board, just needed to customize the kernel a bit, haven't been testing asterisk on that yet, but i definatly will in the sooner future. Kind regards, Martin List-Petersen martin (at) list (dash) petersen (dot) net best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] HFC ISDN card with bristuff from jung hanns.n et?
Hello Robinson, Thursday, June 17, 2004, 1:19:12 PM, you wrote: RTW Hi Alessio RTW Yes, the problems you report do seem similar to the issues RTW I had. I found on the Dells that the audio prompts were very RTW choppy and played slower than normal. Occasionally there would RTW be 'bursts' oav a second or so of 'good' audio. RTW I also suspected IRQ issues but the Dell Mobos had no way RTW of adjusting them. Best thing is to try and get the card on its RTW own unshared IRQ. If this fails, you either have to try a RTW different pc, or collect 600 euros together and send them to RTW Junghanns.net, and they will send you a quadBRI card that does RTW not have this problem. Well card has his own irq, I will try to tweak bios parameters to see if something gets better. Meanwhile since I orderer 2 dell's yesterday hoping to solve the problem I'm going to bang my head against the wall until they arrive Tnx for now ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SFTP
Im having problems with a new install of Asterisk (I had to reinstall because hard drive failed). Ive used debian net install this time and for some reason WS FTP will not connect using SFTP (it keeps coming back with username and password fail) but when I use Putty to connect with the same password and username it works no problems. Any thoughts? Any other programs I can use for SFTP? Cheers, Dean
Re: [Asterisk-Users] embedded Asterisk
On Thu, 17 Jun 2004, listas iPfone wrote: 1. burn the rescue iso mount -o loop -t iso9660 /file /mnt/loop 1. copy the rescue disk to a hard drive cp -dpR /mnt/loop/* /new/location 2. compile asterisk make PREFIX=/new/location install (check if asterisk don't copy all development non-sence) 3. copy all to the flash disk fdisk /dev/hdX[0-9] make partitions mkfs.ext2 /dev/hdX[0-9] mount -t ext2 /dev/hdX[0-9] /mnt/flash cp -dpR /new/location /mnt/flash It is that simple? Probably you want something that actually boots the system too. I don't know if the ISOLINUX pakage supports a LILO kind of thing, but I guess it does. That should be in the MBR of your flash disk and you could probably boot it. I wrote the instructions by mind, so probably something is missing :) Stefan Miklos - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 17, 2004 5:11 AM Subject: Re: [Asterisk-Users] embedded Asterisk Hi, Actually, you the Geode CPU mentioned below is a 5x86 (486 platform) at 233 MHz. If you take Pebble (http://www.nycwireless.net/pebble/), which is a downstripped Debian ( 64 MB) on a readonly ext2 filesystem, you should be grand. Installing asterisk + some extra stuff will probably require, that you have at least a 128MB or 256MB flash or so. Dont go for stripped down but complete distributions which include a lot of stuff that you dont need, e.g. gcc. Go for a rescue system, like i used the SuSE rescue system (14 mb), then you can add what you need (sshd,...) and compile asterisk on another box and then just copy it. My compressed ramdisk image is 32 mb, including all voice prompts and some mp3s for MOH. There are actually quite some board around on that CPU, like Soekris, pcengines and i think also Mikrotik at prices from 120EUR and up. I just put together the demo system for Linuxtag: - Via EPIA 5000 (C3-533), EUR 80,- - Morex case with external power supply, EUR 80,- - some old 256 mb SDRAMM - 128 MB USB memory stick, EUR 30,- - 1 quadBRI (could also easily handle an octoBRI, or a PRI card, with the dual riser pci card you can use 2 cards) The C3-533 is an i586 CPU. According to show translation it needs 30 ms for transcoding 1 channel from g711 to gsm (and vice versa). So, neglecting any overhead caused by channel handling it could transcode 30 channels to gsm. Linux BIOS has support for the EPIA boards, so you can speed up booting very much and also disable the VGA port (very useful for production deployments). I'm running pebble on a pcengines board, just needed to customize the kernel a bit, haven't been testing asterisk on that yet, but i definatly will in the sooner future. Kind regards, Martin List-Petersen martin (at) list (dash) petersen (dot) net best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LDAP synchronization script
Yeah that 'old' message discribes VERY MUCH what I'm doing at the moment. Though there should be an 'application' part and an universal 'user' part. For example the meetme is application specific, should be in the Asterisk tree. But the extentions should basically be templates part of the Asterisk tree which can be used in the universal 'user' part. The what belongs were is my big question at the moment and I personally don't want to design anything LDAP-ish that would become my private tree instead of defacto implementation. Stefan On Thu, 17 Jun 2004, David Hajek wrote: I think I'll use something from this article - http://www.marko.net/asterisk/archives/0205/0006.html -David -Original Message- From: Stefan de Konink [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 1:12 PM To: David Hajek Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LDAP synchronization script I'm planning to incorporate this (native and dynamic) LDAP for my own system on short term. Do you have any LDAP design in mind? Stefan On Thu, 17 Jun 2004, Jeremy Jones wrote: David Hajek Sent: Thursday, June 17, 2004 2:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] LDAP synchronization script Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Not aware of any scripts like that, but... you could use the odbc support in asterisk in conjunction with some slick odbc-ldap connectivity. Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with PRI with T410 messages
Now what is the normal behavior and how can I set it so that * behaves normally? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: Asterisk-Users digest, Vol 1 #4186 - 11 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.n et? (Alessio Focardi) 2. RE: LDAP synchronization script (Stefan de Konink) 3. Re: Problems with PRI with T410 messages (CW_ASN) 4. RE: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from jung hanns.n et? (Robinson Tim-W10277) 5. RE: LDAP synchronization script (David Hajek) 6. Zapata.conf Signaling for Bulgaria (PSTN: Siemens PABX) (Miroslav Nachev) 7. Re: embedded Asterisk (listas iPfone) 8. Re[4]: [Asterisk-Users] HFC ISDN card with bristuff from jung hanns.n et? (Alessio Focardi) 9. SFTP (Dean Collins) 10. Re: embedded Asterisk (Stefan de Konink) --__--__-- Message: 1 Date: Thu, 17 Jun 2004 13:18:51 +0200 From: Alessio Focardi [EMAIL PROTECTED] To: Robinson Tim-W10277 [EMAIL PROTECTED], [EMAIL PROTECTED] Subject: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.n et? Reply-To: [EMAIL PROTECTED] Hello Robinson, Thursday, June 17, 2004, 12:42:21 PM, you wrote: RTW Please can you explain in more details as to what your RTW problem is? I have 2 cards working in one PC, but have had RTW problems with Dell motherboards. voice is out of sync, it syncs for some second if I run something over another console, like, for instance a find / then slips away again. I suspect an Irq problem, what do you think ? What kind of problems have you found with dell's ? Tnx for the help ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] --__--__-- Message: 2 Date: Thu, 17 Jun 2004 13:12:25 +0200 (CEST) From: Stefan de Konink [EMAIL PROTECTED] To: David Hajek [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LDAP synchronization script Reply-To: [EMAIL PROTECTED] I'm planning to incorporate this (native and dynamic) LDAP for my own system on short term. Do you have any LDAP design in mind? Stefan On Thu, 17 Jun 2004, Jeremy Jones wrote: David Hajek Sent: Thursday, June 17, 2004 2:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] LDAP synchronization script Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Not aware of any scripts like that, but... you could use the odbc support in asterisk in conjunction with some slick odbc-ldap connectivity. Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 3 From: CW_ASN [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with PRI with T410 messages Date: Thu, 17 Jun 2004 08:13:03 -0300 Reply-To: [EMAIL PROTECTED] This is a problem I pointed out to Digium a while back, but I am not sure Markster understood the issue and I didn't really have the time to follow it up. It does need fixing though, as it is a major drawback in the current architecture. Rgds Tim Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on and the phone is ringing (which is not the case )and after it will send a RELEASE message saying that the line is busy or the # is invalid .is there any way * can send a progress message instead of the alerting message until it gets the correct message from SER? Thanks Habiyakare Aimable Call Proceeding can be sent only by transit network, not by the local switch or pbx. AFAIK, * behavior for this scenario is
RE: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from jung hanns.n et?
First of all, this is all too vague information you guys are providing here. When you state problems like this, you should be more specific. A) What card are you using (there are lots of HFC-S cards out there). B) What distribution, asterisk-version (stable, HEAD, what date if HEAD) are you using). C) Maybe a output of lspci -v could help D) What kernel are you using ? is ACPI enabled ? E) What configuration are you using bristuff in ? TE mode ? NT mode ? both ? and a lot of other information. The last two mails were basically useless to solve or troubleshoot anything. To say it quite frank, beyond some minor issues with channel assignments and incomming calls from IAX to ZAP (zaphfc, NT mode, Typhoon ISDN card) i do not have problems with bristuff (0.0.2, CVS HEAD 20040324, libpri 20040510). I'm running the whole thing on a Dell Dimension 8300 on Debian Sarge (testing, partly unstable), kernel 2.4.26, with SMP enabled and hyperthreading enabled on the cpu. I could add lots of other information here and the only thing why i'm not reporting the IAX trouble i have is because i currently haven't tested if the same problem persists with CVS stable (which is what kapejod assumes when he develops bristuff) or the newest CVS HEAD (which would require me to change some things in the patches kapejod provides). Please try again, if you would like to have anybody look at your problems. The way you stated your problems until now gives nobody a clue. Kind regards, Martin List-Petersen martin (at) list (dash) petersen (dot) net On Thu, 2004-06-17 at 12:19, Robinson Tim-W10277 wrote: Hi Alessio Yes, the problems you report do seem similar to the issues I had. I found on the Dells that the audio prompts were very choppy and played slower than normal. Occasionally there would be 'bursts' oav a second or so of 'good' audio. I also suspected IRQ issues but the Dell Mobos had no way of adjusting them. Best thing is to try and get the card on its own unshared IRQ. If this fails, you either have to try a different pc, or collect 600 euros together and send them to Junghanns.net, and they will send you a quadBRI card that does not have this problem. Rgds Tim -Original Message- From: Alessio Focardi [mailto:[EMAIL PROTECTED] Sent: 17 June 2004 12:19 To: Robinson Tim-W10277; [EMAIL PROTECTED] Subject: Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.n et? Hello Robinson, Thursday, June 17, 2004, 12:42:21 PM, you wrote: RTW Please can you explain in more details as to what your problem is? RTW I have 2 cards working in one PC, but have had problems with Dell RTW motherboards. voice is out of sync, it syncs for some second if I run something over another console, like, for instance a find / then slips away again. I suspect an Irq problem, what do you think ? What kind of problems have you found with dell's ? Tnx for the help ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SFTP
Your WSFTP program may only have SSH1 but your Debian server may only have SSH2. Look in /etc/ssh/sshd_config Make sure you have 'Protocol 1' I do not recommend this setting as it is not secure. I use F-Secure SSH Client w/Debian and like it. TL P.S. Please take this question to a debian or wsftp support list if this suggestion doesnot solve your problem. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Dean CollinsSent: Thursday, June 17, 2004 7:36 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] SFTP Im having problems with a new install of Asterisk (I had to reinstall because hard drive failed). Ive used debian net install this time and for some reason WS FTP will not connect using SFTP (it keeps coming back with username and password fail) but when I use Putty to connect with the same password and username it works no problems. Any thoughts? Any other programs I can use for SFTP? Cheers, Dean
Re: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.net?
has anyone succesfully installed such scenario ? Yes, see just my e-mail from today. It's in the mailing list archive, see http://lists.digium.com/pipermail/asterisk-users BTW: it's always good to check mailing list archives :-) I'm having problem with Award bios mb pc's... it do works with others, what's your idea ? If it works with one BIOS, but not with the other, then it migth be a) an IRQ problems, see cat /proc/interrups b) a mainboard problem (because usually you've to change the mainboard to change the BIOS) In case of a), try disabling built-in peripherals of the board, e.g. the second serial port, usb host etc. That should make IRQs free. You can also try to install the card into a different slot. Tnx ! Pls ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SFTP
Dean, This really has nothing to do with Asterisk. I suspect you'll get better response by posting to a Linux oriented list. Check your distribution vendor's website, as I'm sure they will have links. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Thursday, June 17, 2004 7:36 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] SFTP Im having problems with a new install of Asterisk (I had to reinstall because hard drive failed). Ive used debian net install this time and for some reason WS FTP will not connect using SFTP (it keeps coming back with username and password fail) but when I use Putty to connect with the same password and username it works no problems. Any thoughts? Any other programs I can use for SFTP? Cheers, Dean
RE: [Asterisk-Users] Problems with PRI with T410 messages
I do not believe you are correct. We see CALL PROCEEDING in both directions as part of the normal ISDN call setup process. See trace below. Asterisk sends 'CALL PROCEEDING' followed immediately by 'ALERTING'. CALL PROCEEDING is normally an acknowledgement to a SETUP. See Q931 below: 3.1.2 CALL PROCEEDING This message is sent by the called user to the network or by the network to the calling user to indicate that requested call establishment has been initiated and no more call establishment information will be accepted. See Table 3-3. ALERTING has a very specific meaning: 3.2.1 ALERTING This message is sent by the called user to the network to indicate that called user alerting has been initiated. See Table 3 23. i.e. the channel to the called party has been established, and the phone at the other end is physically ringing or making some other indication that an incoming call is there to be answered. It is 'ALERTING' that is being sent in the wrong place, as Asterisk sends 'ALERTING' before the remote party (be it a SIP or IAX channel) is actually 'ringing'. Receipt of 'ALERTING' from the called party is the trigger for the calling party to be presented with 'ringback tone'. So to send a 'RELEASE' message with 'busy' after the caller has been told the phone is ringing is not a logical thing to do, and causes a lot of problems here. It needs fixing Rgds Tim Connected to Asterisk CVS-D2004.05.25.23.00.00-06/14/04-12:46:31 currently running on localhost (pid = 4875) mote UNIX connection Protocol Discriminator: Q.931 (8) len=40 Call Ref: len= 1 (reference 1/0x1) (Originator) Message type: SETUP (5) Sending Complete (len= 4) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10) Ext: 1 Progress Description: Call is not end-to-end ISDN; further call progress information may be available inband. (1) ] Calling Number (len=18) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Called Number (len= 5) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '14' ] -- Making new call for cr 1 -- Processing Q.931 Call Setup -- Processing IE 33 (Sending Complete) -- Processing IE 4 (Bearer Capability) -- Processing IE 24 (Channel Identification) -- Processing IE 30 (Progress Indicator) -- Processing IE 108 (Calling Party Number) -- Processing IE 112 (Called Party Number) Protocol Discriminator: Q.931 (8) len=7 Call Ref: len= 1 (reference 129/0x81) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] Protocol Discriminator: Q.931 (8) len=7 Call Ref: len= 1 (reference 129/0x81) (Terminator) Message type: ALERTING (1) Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] -- Executing Wait(Zap/1-1, 2) in new stack -- Accepting call from '0044125679' to '14' on channel 1, span 1 -- Executing Goto(Zap/1-1, default|8714|1) in new stack -- Goto (default,8714,1) -- Executing SetMusicOnHold(Zap/1-1, default) in new stack -- Executing Answer(Zap/1-1, ) in new stack Protocol Discriminator: Q.931 (8) len=11 Call Ref: len= 1 (reference 129/0x81) (Terminator) Message type: CONNECT (7) Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B1 channel ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Executing SayDigits(Zap/1-1, 0044125679) in new stack -- Playing 'digits/0' (language 'en') Protocol Discriminator: Q.931 (8) len=4 Call Ref: len= 1 (reference 1/0x1) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CW_ASN Sent: 17 June 2004 12:13 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problems with PRI with T410 messages This is a problem I pointed out to Digium a while back, but I am not sure Markster understood the issue and I didn't really have the time to follow it up. It does need fixing though, as it is a major drawback in the current architecture. Rgds Tim Hi all, I
RE: [Asterisk-Users] SFTP
You could use winscp3 (it comes from the putty family). It has support for scp and sftp. On Thu, 2004-06-17 at 08:11, Todd Lieberman wrote: Your WSFTP program may only have SSH1 but your Debian server may only have SSH2. Look in /etc/ssh/sshd_config Make sure you have 'Protocol 1' I do not recommend this setting as it is not secure. I use F-Secure SSH Client w/Debian and like it. TL P.S. Please take this question to a debian or wsftp support list if this suggestion does not solve your problem. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dean Collins Sent: Thursday, June 17, 2004 7:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SFTP Im having problems with a new install of Asterisk (I had to reinstall because hard drive failed). Ive used debian net install this time and for some reason WS FTP will not connect using SFTP (it keeps coming back with username and password fail) but when I use Putty to connect with the same password and username it works no problems. Any thoughts? Any other programs I can use for SFTP? Cheers, Dean -- Pablo Endres [EMAIL PROTECTED] ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] HFC ISDN card with bristuff from junghanns.net?
HS a) an IRQ problems, see cat /proc/interrups HS b) a mainboard problem (because usually you've to change the mainboard to HS change the BIOS) HS In case of a), try disabling built-in peripherals of the board, e.g. the HS second serial port, usb host etc. That should make IRQs free. You can HS also try to install the card into a different slot. I'm pretty sure that the card sits on his own IRQ, anyway I'm going to double check that. I'm running fedora core 1 and asterisk was installed using the script I found in the bristuff 0.0.0.2 package. The problem shows in NT or TE mode, the same hard disk installed on a different pc (with another bios) do work. Also I have verified that the hfc card works perfectly using isdn4linux driver. In the motherboard I can tweak PCI LATENCY TIMER: actualy 64 IRQ MODE: actualy APIC also I have tried with hdparm, setting dma mode 3 and other parameters ... still nothing ! Tnx for the help ! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SFTP
Filezilla SFTP, FTP, SSL-FTP Works on every linux distro I've tried as well as cygwin and several other encrypted file transfer servers both Win32 and Unix-based http://filezilla.sf.net/ MATT--- -Original Message-From: Todd Lieberman [mailto:[EMAIL PROTECTED]Sent: Thursday, June 17, 2004 8:12 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] SFTP Your WSFTP program may only have SSH1 but your Debian server may only have SSH2. Look in /etc/ssh/sshd_config Make sure you have 'Protocol 1' I do not recommend this setting as it is not secure. I use F-Secure SSH Client w/Debian and like it. TL P.S. Please take this question to a debian or wsftp support list if this suggestion doesnot solve your problem. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Dean CollinsSent: Thursday, June 17, 2004 7:36 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] SFTP Im having problems with a new install of Asterisk (I had to reinstall because hard drive failed). Ive used debian net install this time and for some reason WS FTP will not connect using SFTP (it keeps coming back with username and password fail) but when I use Putty to connect with the same password and username it works no problems. Any thoughts? Any other programs I can use for SFTP? Cheers, Dean
RE: [Asterisk-Users] Asterisk-Users List Etiquette
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Lee Sent: Tuesday, June 15, 2004 6:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk-Users List Etiquette Kevin Walsh wrote: Steven Critchfield [EMAIL PROTECTED] wrote: You forgot to add in how awful it is when people post using HTML and then override font sizes or assume blue is an appropriate font color for their message. While I know some people don't like it when I turn my attention to them, if it takes me even one more button press to be able to read your mail, it isn't likely to be interesting to me to even bother helping you with your problem. Since the majority of unix users understand how each of us tweak our environment to be the most productive for us, we don't like it when you take liberties with our settings. He also forgot to mention how awful it is when people lazily top-post instead of taking the time to format their followups correctly. This is especially true when trying to follow a thread found in the archives. I fully agree with your anti-HTML comments, by the way. I think you will find that about half the people out there disagree with this sentiment (a guess based on the number of top and bottom posters I have seen) so no matter how often you ask it is not likely to change things much. Top posting is what a lot of people are very comfortable with. It also has the advantage in lists that when you step through a thread the answer to the last item is ready for you to read. So If you bottom post you make life harder for the thread reader but if you top post you make life harder for those that get a long mail out of the archives.Who should we favor? Don't ask why I am bottom posting, I have no good reason, it just so happens that I am. I don't like HTML either but a lot of people don't know they can switch it off or that it even exists (its a word processor isn't it?). Getting offended by these personal preferences just leads to that etiquette problem, the god ol flame war. Or at least heated debate that will never be won with so many advocates for each side, that the lists become quite full of top/bottom html/text arguments. Please don't bring these subjects into things it just makes people with other views upset. I'm quite content to post at the top, bottom, or inline. It really just depends on the nature of the message I'm replying to, the subject, context, and format of earlier messages in the thread. However, my preference is for top posting. The reason, is that in order to read my message here, you had to scroll through ~70 lines of previous discussion. Stuff that you've /already/ read since you've been following this thread. Oh! Wait, you found this in an archive, so you /want/ to have the thread fully quoted so you don't have to go hunting down the references. Good, that's why I didn't trim this post. Oh, wait, the guys that are following this thread as it's being discussed would prefer that I trim out the stuff up there, in which case, I would be neither top posting, nor bottom posting. This message would be a post unto itself that wouldn't have any quoted material at all. Afterall, you've already read the referenced material. So, the bottom line is that top-posters are lazy? I say yes, we are. We don't want to have to scroll through pages of quoted material just to get to the new stuff. I say that the bottom posters are lazy. They want a bottom post so that they enter into a thread 12 messages later, and not have to read the thread 'backwards.' Read your mail to begin with, and you wouldn't have this problem, and you would actually start to appreciate the top posters, because they're making it so you don't have to scroll through ~70 lines of quoted material to get to the new stuff. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk-Users List Etiquette
A: Because we read the question in the previous message. Q: Why should I post my reply above the quoted text? -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Wednesday, June 16, 2004 2:39 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk-Users List Etiquette On Wed, 16 Jun 2004, Nicholas Bachmann wrote: You might try reading http://www.caliburn.nl/topposting.html -- it explains why people don't like top posting. Or read this quote: A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SFTP
Matt, thanks for your suggestion another kind soul just suggested it about 10 minutes ago and it is already working like a charm, as for those that dont think this is an asterisk problem phooey ;) Night all. Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Thursday, 17 June 2004 11:25 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] SFTP Filezilla SFTP, FTP, SSL-FTP Works on every linux distro I've tried as well as cygwin and several other encrypted file transfer servers both Win32 and Unix-based http://filezilla.sf.net/ MATT--- -Original Message- From: Todd Lieberman [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 8:12 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SFTP Your WSFTP program may only have SSH1 but your Debian server may only have SSH2. Look in /etc/ssh/sshd_config Make sure you have 'Protocol 1' I do not recommend this setting as it is not secure. I use F-Secure SSH Client w/Debian and like it. TL P.S. Please take this question to a debian or wsftp support list if this suggestion doesnot solve your problem. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Dean Collins Sent: Thursday, June 17, 2004 7:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SFTP Im having problems with a new install of Asterisk (I had to reinstall because hard drive failed). Ive used debian net install this time and for some reason WS FTP will not connect using SFTP (it keeps coming back with username and password fail) but when I use Putty to connect with the same password and username it works no problems. Any thoughts? Any other programs I can use for SFTP? Cheers, Dean
RE: [Asterisk-Users] Asterisk-Users List Etiquette
On Thu, 2004-06-17 at 09:21 -0400, Troy Settle wrote: ..snip.. However, my preference is for top posting. The reason, is that in order to read my message here, you had to scroll through ~70 lines of previous discussion. Stuff that you've /already/ read since you've been following this thread. ..snip..} Sorry to butt into this thread, but I think this is where you went wrong. There was absolutely no need to quote 70+ lines of text to say what you had to say. You're supposed to quote the relevant bits (as I did with this email), not the entire thread. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failed to authenticate on INVITE
Hi Jason, Thanks for your reply. I didn't really want to use the insecure option, that defeats the purpose of using a password :) I was, however, able to specify user= in my sip.conf entity and that solved the problem I was having. Thanks again. - Eric On Thu, 17 Jun 2004 10:17:54 +0100 Jason Williams [EMAIL PROTECTED] wrote: At 16:49 16/06/2004 -0400, Eric wrote: I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error Failed to authenticate on INVITE trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. include the line in sip.conf for each user the call insecure=yes ; To match a peer based by IP address only and not peer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-Users List Etiquette
On Thursday 17 June 2004 09:21, Troy Settle wrote: However, my preference is for top posting. The reason, is that in order to read my message here, you had to scroll through ~70 lines of previous discussion. Stuff that you've /already/ read since you've been following this thread. That's because you didn't trim anything. To see what I wrote to you You had less than 10 lines to look at. Please don't use absurdity to try and prove your point. Oh! Wait, you found this in an archive, so you /want/ to have the thread fully quoted so you don't have to go hunting down the references. Good, that's why I didn't trim this post. Um no, that's why the archives are threaded themselves. Attempt at reductio ad dbsurdum #2 failed. Oh, wait, the guys that are following this thread as it's being discussed would prefer that I trim out the stuff up there, in which case, I would be neither top posting, nor bottom posting. This message would be a post unto itself that wouldn't have any quoted material at all. Afterall, you've already read the referenced material. I consider trimming the quoted text and replying to the bits you keep as they occur bottom posting -- your text is FOLLOWING the relevant bits of the conversation. Inline posting is something completely different and it's even more heinous: So, the bottom line is that top-posters are lazy? [ yes, they are absolutely. Inline posters are even worse! ] I say yes, we are. We don't want to have to scroll through pages of quoted material just to get to the new stuff. [ so trim your damned posts ] That above is an example of inline posting. Some managers have a penchant for that. I say that the bottom posters are lazy. They want a bottom post so that they enter into a thread 12 messages later, and not have to read the thread 'backwards.' Read your mail to begin with, and you wouldn't have this problem, and you would actually start to appreciate the top posters, because they're making it so you don't have to scroll through ~70 lines of quoted material to get to the new stuff. That's not laziness, that is following natural language laws. I have over 25k messages in my local copy of asterisk-users. My MUA understands message threading so if people posted the One True Way (editing quoted content and replying underneath, as I am doing to you here) then there is no problem following the flow of the thread, and if I need more information I move up to the message parent and see the entire message. It's not a difficult thing to understand, and this absurdity you're spewing to try and prove your point only goes to show that your argument doesn't hold much logic. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk on FreeBSD
Hello, ereyone! I have just installed Asterix on my FreeBSD (-current) box I'm planning to use it as H323 PBX for softphones Currently I'm stuck in transfering a call to another machine running H323 client When I define forwarding address as H323/ip$192.168.1.77|20|r Asterisk will crash immediately with Segmentation Fault when trying to transfer Program received signal SIGSEGV, Segmentation fault. 0x289f1314 in _init () from /usr/local/lib/asterisk/modules/chan_h323.so (gdb) x/3i $eip 0x289f1314 _init+12668: cmpb $0x0,(%eax) 0x289f1317 _init+12671: je 0x289f1327 _init+12687 0x289f1319 _init+12673: sub$0xc,%esp When I define call forwarding address as: H323/ip$192.168.1.77/|20|ri.e. additional / after IP It will perfectly connect and transfer call if there is H323 cli running If target machine is powered off or no software is running it will behave weird It will eat 100% cpu, hang forever and transmit silence to caller However tracing h.323 shows that it indeed detects that there is no H.323 connection to target avaible -- PBX1 is calling host ip$192.168.1.77 -- Call token is ip$localhost/25892 -- Call reference is 25892 -- Called ip$192.168.1.77 -- No phone running for ip$192.168.1.77:1720 == H.323 Connection deleted. Any help will be much appreciated. I will be glad to provide any required debuggin info, etc. Cheers, AL. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD
On Thu, Jun 17, 2004 at 05:28:00PM +0400, AK wrote: Hello, ereyone! I have just installed Asterix on my FreeBSD (-current) box I'm planning to use it as H323 PBX for softphones Currently I'm stuck in transfering a call to another machine running H323 client When I define forwarding address as H323/ip$192.168.1.77|20|r Asterisk will crash immediately with Segmentation Fault when trying to transfer Program received signal SIGSEGV, Segmentation fault. 0x289f1314 in _init () from /usr/local/lib/asterisk/modules/chan_h323.so (gdb) x/3i $eip 0x289f1314 _init+12668: cmpb $0x0,(%eax) 0x289f1317 _init+12671: je 0x289f1327 _init+12687 0x289f1319 _init+12673: sub$0xc,%esp Just to clarify things for people, %eax is set to NULL (so its a null pointer dereference). When I define call forwarding address as: H323/ip$192.168.1.77/|20|ri.e. additional / after IP It will perfectly connect and transfer call if there is H323 cli running If target machine is powered off or no software is running it will behave weird It will eat 100% cpu, hang forever and transmit silence to caller However tracing h.323 shows that it indeed detects that there is no H.323 connection to target avaible -- PBX1 is calling host ip$192.168.1.77 -- Call token is ip$localhost/25892 -- Call reference is 25892 -- Called ip$192.168.1.77 -- No phone running for ip$192.168.1.77:1720 == H.323 Connection deleted. Any help will be much appreciated. I will be glad to provide any required debuggin info, etc. Cheers, AL. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk-Users List Etiquette
-Original Message- From: Gonzalo Servat Sent: Thursday, June 17, 2004 9:34 AM Sorry to butt into this thread, but I think this is where you went wrong. There was absolutely no need to quote 70+ lines of text to say what you had to say. You're supposed to quote the relevant bits (as I did with this email), not the entire thread. It's an open mailing list, you're not butting in at all. I agree with you completely, however, there is this great tool called 'exageration' that is sometimes used to make a point when a real-world example would be too small to be perceived as signifigant. For those nay-sayers, please look at my post carefully. I bottom posted, keeping the existing style, and while I left the quoted material untrimmed, I also mentioned the other extreme, which is to completely exclude any quoted material at all. The bottom line of this issue is that everyone has their preferences, and no amount of crying and whining will cause the other side to comply with your wishes. There are valid reasons for both posting styles, live with it. Those who continue to whine and cry about top posting need to be larted with a vengence. It's like the last cry of those who lost the vi-vs-emacs debate. Just because you prefer one over the other doesn't make everyone else 'wrong.' IMO, the top-vs-bottom topic really needs to be classified right along side with the RH-vs-Debian, red-vs-blue, unix-vs-windows, ford-vs-chevy, linux-vs-bsd, and other similar cases of personal preferences. The is no winner, there never will be a winner. BTW, for those of you who are curious, I too dispise HTML formatted email in a mailing list environment. I also dislike those who flagrantly disregard existing styles within a thread (but, it's ok if different threads have different styles). I also have very low regard for those among us who would hijack a thread. I don't use a threaded mail reader myself (sucks to be me), but when browsing archives by thread, it's really annoying to find questions about personal lubricant in the middle of a heated debate about top-vs-bottom. Of course, sometimes a thread will mutate naturally, at which point, it may be appropriate to change the subject (which I'm not going to do, since I'm too damned lazy. Oh, for those curious, my single, biggest beef with mailing lists, is the inclusion of a list tag in the Subject: line. I know it's Asterisk-Users, because it says so in the To: line. It also says so in the List-ID: and Sender: lines. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-addons compilation error
Folks I am getting the following error as of today after updating both asterisk and asterisk-addons. These are both under /usr/src. Any ideas? dora-debian:/usr/local/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: error: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: error: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: error: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: error: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 dora-debian:/usr/local/src/asterisk-addons# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk-Users List Etiquette
What is this? Day Three? What is the standing record on this list for flame wars? You guys need to do a sanity check. These posts are nothing more than SPAM and Ive just added to it. I feel so dirty now. [EMAIL PROTECTED] 6/17/2004 9:08:09 AM -Original Message- From: Gonzalo Servat Sent: Thursday, June 17, 2004 9:34 AM Sorry to butt into this thread, but I think this is where you went wrong. There was absolutely no need to quote 70+ lines of text to say what you had to say. You're supposed to quote the relevant bits (as I did with this email), not the entire thread. It's an open mailing list, you're not butting in at all. I agree with you completely, however, there is this great tool called 'exageration' that is sometimes used to make a point when a real-world example would be too small to be perceived as signifigant. For those nay-sayers, please look at my post carefully. I bottom posted, keeping the existing style, and while I left the quoted material untrimmed, I also mentioned the other extreme, which is to completely exclude any quoted material at all. The bottom line of this issue is that everyone has their preferences, and no amount of crying and whining will cause the other side to comply with your wishes. There are valid reasons for both posting styles, live with it. Those who continue to whine and cry about top posting need to be larted with a vengence. It's like the last cry of those who lost the vi-vs-emacs debate. Just because you prefer one over the other doesn't make everyone else 'wrong.' IMO, the top-vs-bottom topic really needs to be classified right along side with the RH-vs-Debian, red-vs-blue, unix-vs-windows, ford-vs-chevy, linux-vs-bsd, and other similar cases of personal preferences. The is no winner, there never will be a winner. BTW, for those of you who are curious, I too dispise HTML formatted email in a mailing list environment. I also dislike those who flagrantly disregard existing styles within a thread (but, it's ok if different threads have different styles). I also have very low regard for those among us who would hijack a thread. I don't use a threaded mail reader myself (sucks to be me), but when browsing archives by thread, it's really annoying to find questions about personal lubricant in the middle of a heated debate about top-vs-bottom. Of course, sometimes a thread will mutate naturally, at which point, it may be appropriate to change the subject (which I'm not going to do, since I'm too damned lazy. Oh, for those curious, my single, biggest beef with mailing lists, is the inclusion of a list tag in the Subject: line. I know it's Asterisk-Users, because it says so in the To: line. It also says so in the List-ID: and Sender: lines. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cheap (US$120 or less) SIP Phones
On 17 June 2004 Eric Wieling wrote: These are the three cheap SIP phones that I've used. Grandstream BT10x $65/street Number only LCD Zultys ZIP 2 $100/retail No LCD Uniden UIP 200 $120/retail PoE, built-in switch Are there any online retailers that carry the Uniden UIP series phones? I did a quick Froogle search to no avail. -- Tony Kava Senior Network Administrator Pottawattamie County, Iowa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-addons compilation error
Hi, I posted the same problem yesterday/day b4? Add CFLAGS+=-I../asterisk/include to the top of the Makefile -- [ Nick Luckcuck | [EMAIL PROTECTED] ] [ Junior Software Developer | Motorola ] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Santiago Sent: 17 June 2004 15:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk-addons compilation error Folks I am getting the following error as of today after updating both asterisk and asterisk-addons. These are both under /usr/src. Any ideas? dora-debian:/usr/local/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: error: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: error: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: error: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: error: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 dora-debian:/usr/local/src/asterisk-addons# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote rebooting a Cisco 7940
Ahh, of course :-) A little fiddling around with expect and I can reboot it from a webpage now :-) Thanks. Best Regards Michael On Wed, 16 Jun 2004 14:32:15 -0500 Roger [EMAIL PROTECTED] wrote: Michael Løjtnant wrote: Hi, I have seen a couple of scripts that should be able to remotely reboot the 79xx phones, but I haven't been able to make it work for my 7940. Anyone able to guide me in the right direction? I am running the SIP 7.1 firmware. Telnet to the phone's ip address, enter the password and type reset. Get the phones ip off a 'sip show peers' on asterisk or on the phone hit Settings-3-5 for the ip address. The default password is cisco. - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ebay X101P Card. CRAP!!!
Hi People, I know that this is a Digium forum, and actually i will buy cards now from Digium too. But a have just a question. For test purposes and of course save some money a buy from Ebay a " Mercury M/N: AMI-IA92 card." With this card Asterisk work well - my linux appear like "Tiger Jet card". But i notice some problems: 1 - Dropped calls (I try avoid with Busycount=x) Works but sometimes drops anyway. 2 - Freeze the line. (Sometimes this cards freeze the line and just become normal after restart asterisk). 3 - Voice problems, I notice that this card has some problem with Sound because sound become unstable for the person that listen in the other side. (Appears using a VAD making the call very bad) So anyone has the same problem ? Did anyone know a fix for that ? I will buy a Digium "Official" card just to test and see the diferences. Anyway more info from everyone will be very good. Thanks alot. Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as Internet Talk Radio PBX system
I see in the archives a brief thread between Barton and w
last November 2003 about streaming to the Internet. Id like to use an Asterisk to
mediate multiple VOIP calls originated from the Internet to the studio to be
mixed then passed out to an encoding PC thence back to Internet
{~}
+---+
+---+ +---+ +---+
{ Internet }
| Asterisk |
--- line out - | Mixer |-- | Encoding | | streaming |
{ VOIP Calls}Ethernet--| |
| | | PC | |
Server |
{__}
+---+
- Line in
--- ++ +---+ +---+
|
| |
|
|
?? cd |
|
Talk Show
Callers
Mic
Internet
Via VOIP
Notice there need not be ANY telco POTS lines.
I wonder if there is a group discussion of this type of
functionality.
Would the LINE OUT/IN from Asterisk to
analog MIXER console be PC Sound cards or something more
discrete like a form of telco line cards?
We do not need the additional freq crunching done, typically,
to interface to limited bandwidth telco network..
Jim
Wireless Tech Radio
www.wirelesstechradio.com
I have thought about doing this as well, for what may be thesame application. The easiest way to do it would be to use theConsole channel and audio drivers and use a mixer -- keep inmind, I'm thinking of a radio talk show, presumably with a mixer,other audio sources, etc. It would look something like this: +--+--- line out --+---+ +--+POTS --| Asterisk | | mixer |---| streaming server | +--+-- line in +---+ +--+ | | | | | | CD | | | SIP Clients, Etc. Mic | Internet Etc.Where line out of the Asterisk goes to an input of the mixerand line in is connected to a monitor port on the mixer.This would be very simple to do and wouldn't require conferences.You could map inbound calls to some telephone if you wantedto screen callers or anything like that and then forwardthe call to the console extension when you are ready togo on the air.
RE: [Asterisk-Users] Asterisk-Users List Etiquette
Being new to this list i must tread carefully but Who cares where the answers are so long as they are helpful and to the point. If i ask a question it's just nice to get a good clear and concise answer. Makes no odds to me where the answer is in the reply. Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Kohlsmith Sent: 17 June 2004 14:41 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk-Users List Etiquette On Thursday 17 June 2004 09:21, Troy Settle wrote: However, my preference is for top posting. The reason, is that in order to read my message here, you had to scroll through ~70 lines of previous discussion. Stuff that you've /already/ read since you've been following this thread. That's because you didn't trim anything. To see what I wrote to you You had less than 10 lines to look at. Please don't use absurdity to try and prove your point. Oh! Wait, you found this in an archive, so you /want/ to have the thread fully quoted so you don't have to go hunting down the references. Good, that's why I didn't trim this post. Um no, that's why the archives are threaded themselves. Attempt at reductio ad dbsurdum #2 failed. Oh, wait, the guys that are following this thread as it's being discussed would prefer that I trim out the stuff up there, in which case, I would be neither top posting, nor bottom posting. This message would be a post unto itself that wouldn't have any quoted material at all. Afterall, you've already read the referenced material. I consider trimming the quoted text and replying to the bits you keep as they occur bottom posting -- your text is FOLLOWING the relevant bits of the conversation. Inline posting is something completely different and it's even more heinous: So, the bottom line is that top-posters are lazy? [ yes, they are absolutely. Inline posters are even worse! ] I say yes, we are. We don't want to have to scroll through pages of quoted material just to get to the new stuff. [ so trim your damned posts ] That above is an example of inline posting. Some managers have a penchant for that. I say that the bottom posters are lazy. They want a bottom post so that they enter into a thread 12 messages later, and not have to read the thread 'backwards.' Read your mail to begin with, and you wouldn't have this problem, and you would actually start to appreciate the top posters, because they're making it so you don't have to scroll through ~70 lines of quoted material to get to the new stuff. That's not laziness, that is following natural language laws. I have over 25k messages in my local copy of asterisk-users. My MUA understands message threading so if people posted the One True Way (editing quoted content and replying underneath, as I am doing to you here) then there is no problem following the flow of the thread, and if I need more information I move up to the message parent and see the entire message. It's not a difficult thing to understand, and this absurdity you're spewing to try and prove your point only goes to show that your argument doesn't hold much logic. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk-Users List Etiquette
On Thu, 2004-06-17 at 09:23, Troy Settle wrote: A: Because we read the question in the previous message. Q: Why should I post my reply above the quoted text? You are assuming that everyone subscribed to the list is reading you particular thread. If they're not, but are mostly just skimming through and pausing to read when something looks interesting, top posting makes it hard to understand what the discussion is about. In any case, I'm not ramming this down anyone's throat. If you don't want to top post, then don't. But, I think that after everything is weighed (e.g. people finding threads with top posts on Google 10 years later), top posting would come out the loser. Alrighty, that's enough for me! Two posts about this is all I'm contributing. Kanwar Systems Aligned Inc. www.systemsaligned.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Blank faxes with RxFAX
Hello All,
I have downloaded and installed spandsp and downloaded rxfax, etc and
rebuilt asterisk with app_rxfax. I have added the following to my
extensions.conf:
[macro-faxreceive]
; ${ARG1} - sendto e-mail
exten = s,1,Wait(2)
exten = s,2,Answer
exten =
s,3,SetVar(FAXFILE=/var/spool/asterisk-fax/fax-${MACRO_EXTEN}-${TIMESTAMP})
exten = s,4,SetVar(EMAILADDR=${ARG1})
exten = s,5,rxfax(${FAXFILE}.tif)
exten = s,6,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERIDNAME})
And everything looks like it is running fine, meaning that I don't seem to
get any errors.
However, all the faxes seem to be blank. The only references to anything
like this that I could find previously on the list were in regards to 8-byte
tif files. This doesn't seem to be the same issue, since the files are much
larger (~8k). I am trying to receive the faxes over a PRI running into a
TE405P. Any suggestions? Is anyone using RxFAX successfully in a
configuration like this? If so, what are you doing differently? Any help
would be appreciated.
Thanks,
Patrick
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[Asterisk-Users] Problem with bridging two external lines
We're having a strange problem when an external call is transferred to an external line. Once the transfer happens, the other line gets opened but as far as we can tell the number never gets dialed. The person being transferred gets an extremely loud squealing noise and usually disconnects. Both lines then remain open indefinitely. We've set an absolutetimeout, but it doesn't appear to be working. (It's set to 600 seconds, yet after 15 minutes both lines are still open.) The only way to fix it is to either use a soft hangup or restart the system. The other really strange part is that this appears to be semi-random. Calling in from a land-line and being transferred to a cell phone, for example, usually works fine. But calling in from a cell phone and being transferred to a cell phone results in the problem. This SHOULDN'T make a difference, as both calls are still land line calls until they get to the cell tower, but we've consistently had the problem when going cell to cell. I'm at a complete loss here. Any ideas? -- Alex Malinovich Golden Technologies, Inc. (219) 462-7200 x 216 http://www.golden-tech.com signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] embedded Asterisk
On Jun 17, 2004, at 4:48 AM, Stefan de Konink wrote: It is that simple? Probably you want something that actually boots the system too. I don't know if the ISOLINUX pakage supports a LILO kind of thing, but I guess it does. That should be in the MBR of your flash disk and you could probably boot it. I wrote the instructions by mind, so probably something is missing :) ISOLINUX is part of a family--SYSLINUX for booting from hard drives, ISOLINUX for booting from CDs, and PXELINUX for booting over the network. The configuration is nearly identical for all three. Strictly speaking, you don't really even need the rescue disk. It's surprisingly easy to build a complete Linux system from scratch using uclibc and busybox. Just build busybox statically linked to uclibc (amazingly enough, the last time I did that, the static uclibc busybox was smaller then the dynamically linked glibc busybox) and install it to a temp directory. Then create a couple extra directories (/dev, /tmp, /etc), populate /dev, create a short /etc/passwd and /etc/group, and you should have a bootable Linux image in under 1 MB. Add asterisk to that, and you'll be ready to go. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SJphone regestration problem - Help!
I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address: 192.168.0.1 Caller ID: sip:[EMAIL PROTECTED] Register - checked. Account: markspc Password blank. In SJphone Options-Profiles, when I new create a profile, no matter what Profile type I selected(there three type:Direct SIP Calls, Simple SIP proxy, Calls through SIP Proxy), in the SIP tab, there are only four settings I can set. The four settings are 1 Use application/sip instead of message/sipfrag for Notify bodies 2 expose software version 3 Restrict caller identity(support varies for proxies from different vendors 4 use short headers I installed the SJphone vision 222b on Linux. Is there something simple I missed? or am I on the wrong direction? Help would be greatly greatly appreciated. Thanks Rui ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-Users List Etiquette
On Thursday 17 June 2004 10:21, Simon wrote: If i ask a question it's just nice to get a good clear and concise answer. Makes no odds to me where the answer is in the reply. Precisely -- this is what this mini flame thread is all about. Many of us believe that top posting, not trimming, etc. does NOT provide a clear and concise answer. I tend to agree -- if the top-poster clears away all the crap that 99% of top posters DO NOT clear away, 75% of my beef with top posting would vanish. The unfortunate case is that top posters seem to be inherently lazy, as is evidenced by 99% of them NOT trimming anything in their replies. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems with PRI with T410 messages
I do not believe you are correct. We see CALL PROCEEDING in both directions as part of the normal ISDN call setup process. See trace below. Asterisk sends 'CALL PROCEEDING' followed immediately by 'ALERTING'. CALL PROCEEDING is normally an acknowledgement to a SETUP. See Q931 below: 3.1.2 CALL PROCEEDING This message is sent by the called user to the network or by the network to the calling user to indicate that requested call establishment has been initiated and no more call establishment information will be accepted. See Table 3-3. ALERTING has a very specific meaning: 3.2.1 ALERTING This message is sent by the called user to the network to indicate that called user alerting has been initiated. See Table 3 23. i.e. the channel to the called party has been established, and the phone at the other end is physically ringing or making some other indication that an incoming call is there to be answered. It is 'ALERTING' that is being sent in the wrong place, as Asterisk sends 'ALERTING' before the remote party (be it a SIP or IAX channel) is actually 'ringing'. Receipt of 'ALERTING' from the called party is the trigger for the calling party to be presented with 'ringback tone'. So to send a 'RELEASE' message with 'busy' after the caller has been told the phone is ringing is not a logical thing to do, and causes a lot of problems here. It needs fixing Rgds Tim Tim: Call proceeding is not mandatory in local termination (at least in EuroISDN). Alerting is mandatory (obviously). Some class 5 switches sends Call Proceeding only when the received SETUP will be routed thru CCS or CAS routes, and only when a timer (I can't remember the timer number) expires. The Call Proceeding must be retransmitted to A side. Call Proceeding message is used mostly in transit environments. Obviously, Ringing can't be used when unallocated or busy conditions are detected. The correct procedure for successful call with Call Proceeding and Setup Acknowledge: 1) A-Setup 2) Setup acknowledge -B 3) Call Proceeding -B 4) Ringing -B 5) Answer -B Or 5) Release A-B (by expiration time) The correct procedure for unsuccessful (1 or 17 cause) call without Call Proceeding, with Setup Acknowledge: 1) A-Setup 2) Setup acknowledge -B 3) Release -B (ITU-T release cause i.e.: 1 or 17) As you said, it needs to be fixed. Regards, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk-addons compilation error
Luckcuck Nick-LCKN001 [EMAIL PROTECTED] wrote: I posted the same problem yesterday/day b4? Add CFLAGS+=-I../asterisk/include to the top of the Makefile Alternatively (and IMHO, better), make sure you do make install in asterisk BEFORE trying to do make in asterisk-addons. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Welltech FXO: initial tests
Hi Jorge!Our application rom version is 4fxosip.102boot version is boot.104I think we need to upgrade the app rom to version 103.I get intowelltech ftp server and founda file called 4fxosipN2004_05_17.BIN. Do you know ifthat isthe last version for the 3804?I solvedsome of theproblems I had.1. I can call between the 2 phones with and without reinvite.2. I can call from SIP to pstnIf I call from pstn, the 3804 answer and it dials extension 9 as specified in the bureau table, but it annot dial any internal extension.I hope to solve this last prob with the firmware upgradeMany thanks for you helpBest RegardsClaudio Loletti Email.it, the professional e-mail, gratis per te: clicca qui Sponsor: Lerboristeria.biz: per la tua bellezza e salute il miglior assortimento di prodotti erboristici ed oggettistica online Clicca qui
Re: [Asterisk-Users] 911 emergency service and VoIP
Joe Baptista wrote: I understand that most VoIP providers allow for 911 calling but that 911 service is not the same as that available to PSTN. From what I understand a 911 Call Will Go To A General Access Line at the Public Safety Answering Point (PSAP). This is different from the 911 Emergency Response Center where traditional 911 calls go. Does anyone know how I can get information on howto contact the people at the Public Safety Answering Points (PSAPs)? Is there alist somewhere I can reference. thanks joe baptista Joe, You are slightly confused. Let me explain how it works. When you place a 911 call, it is sent to the 911 selective router at the [I/C]LEC. The 911 selective router does an ALI (Automatic Location Identification) dip against the ANI (Automatic Number Identification) that is present on the call. The ANI is going to be the CallerID number that you/your provider present. When the ALI information is returned to the 911 selective router, it makes the decision which PSAP to send your call to based on the location in the ALI. The call is then routed to the PSAP. The PSAP gets the call and the ANI. They in turn do an ALI dig against the ANI to get the location information on their screens. If no ALI is present in the database for the ANI you're using, the call is default routed to the county PSAP because no positive route can be established without ALI information. When you call 911 without ALI information present, it is 911 service. When you make a call from an ANI that has accurate ALI information, you are using E911 or Enhanced 911 service. If you have PRI service into your * server, it is possible - though not always easy - to set the ALI database information specific for each ANI (DID number) that you use. I do this with our PRI's. Depending on which number we present to the telco, the ALI is different. Now, what you describe might very well be how Vonage and other providers are providing 911 access but, it is most definately NOT even basic 911 as it doesn't go to the PSAP, even the default-route PSAP. It is simply them mapping 911 calls to go to NPA-NXX-NXXX instead. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP wiretapping article
Of course, big brother wants his say in the matter. http://www.wired.com/news/politics/0,1283,63884,00.html?tw=wn_2polihead ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SJphone regestration problem - Help!
Rui, Create a profile (I've used both Simple SIP and calls through SIP), then click on it. This should enable the Initialize button. It opens a window with the fields: Proxy Domain Account Password CallerID I am using SJPHone on windows however. Look around in your Linux SJphone for the initialize option. In SJphone Options-Profiles, when I new create a profile, no matter what Profile type I selected(there three type:Direct SIP Calls, Simple SIP proxy, Calls through SIP Proxy), in the SIP tab, there are only four settings I can set. Use local outbound proxy - checked. Proxy IP Address: 192.168.0.1 Caller ID: sip:[EMAIL PROTECTED] Register - checked. Account: markspc Password blank. Ty Purcell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk as Internet Talk Radio PBX system
Notice there need not be ANY telco POTS lines. I wonder if there is a group discussion of this type of functionality. Would the LINE OUT/IN from Asterisk to analog MIXER console be PC Sound cards or something more discrete like a form of telco line cards? We do not need the additional freq crunching done, typically, to interface to limited bandwidth telco network.. Jim Wireless Tech Radio www.wirelesstechradio.com This will work fine with a regular (but DUPLEX) soundcard, i.e. most $40-50 soundcards, but based on your mixer, you may need some matchboxes that do impedance and level matching for you. The soundcard in most pc's (unless you spend big $$$ for a pro one, which I recommend against) will have levels that are too high for your pro gear. You might get away with just padding the input and bringing down the gain on the mixer input. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to let users change Voice Mail password in Asterisk
On Thu, 2004-06-17 at 08:20 -0700, Deepak Malhotra wrote: Hello Any idea or code on How to allow users to change their voice mail password over the Phone. The only way io know is to change in voicemail.conf file and restart asterisk. Try dialing your voicemail extension, enter your password, then press 0, then press 4. Follow the prompts. HTH. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 emergency service and VoIP
On Thursday 17 June 2004 11:38, John Fraizer wrote: If you have PRI service into your * server, it is possible - though not always easy - to set the ALI database information specific for each ANI (DID number) that you use. I do this with our PRI's. Depending on which number we present to the telco, the ALI is different. Do you have information on how to do this? This is *precisely* what I want to do. I assumed you set this up with your telco and then set the caller ID to the # matching the address you wanted, leaving the telco to do the address match. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as Internet Talk Radio PBX system
To use Asterisk as platform for such a system you probably want to have a Alsa enabled card which supports routing of multiple channels in and out. So Asterisk is like the intermediate 'engine' that routes the signal. (Or sort some soft-mixer). A user is then placed in a Meetme room and the hold signal would be the live show? OUTPUT||==o || Soundcard 1 (studio/broadcast) =||=|---\ || Soundcard 2 (prelisten/desk) |\ | | | | SIP/IAX incomming|/ | || Meetme |---/ | Meetme | | Meetme | Technical picks of the phone by prelistening, transfers it to a new or existing meetme. When the actual 'meeting' starts, the meetme gets in the air studio audio (presenter) gets in by the Alsa interface. Somebody earlier suggested Asterisk for use in remote broadcasts (on a location for example). With two boxes and a ISDN line on both sides, some encoding and you are in business too. Asterisk as a application platform is quite powerfull, but probably has some overhead which 'all-in-one' products have. Stefan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Resend to correct graphic - Internet Talk Radio use Talk Show PBX
I see in the archives a brief thread between Barton and w last November
2003 about streaming to the Internet. I'd like to use an Asterisk to
mediate multiple VOIP calls originated from the Internet to the studio
to be mixed then passed out to an encoding PC thence back to Internet
{~} +++--+ +-+ +-+
{Internet } |Asterisk|--line out-|Mixer |--|Encoding |-|streaming|
{VOIP }-||| | | PC| | Server |
{_} ++ -Line in--+--+ +-+ +-+
| | ||
|?? cd ||
Talk Show Callers Mic Internet
Via VOIP
Notice there need not be ANY telco POTS lines.
I wonder if there is a CURRENT group discussion of this type of
functionality.
Would the LINE OUT/IN from Asterisk to analog MIXER console be PC
Sound cards or something more discrete like a form of telco line cards?
We do not need the additional freq crunching done, typically, to
interface to limited bandwidth telco network..
Jim
Wireless Tech Radio
www.wirelesstechradio.com
---
I have thought about doing this as well, for what may be the
same application. The easiest way to do it would be to use the
Console channel and audio drivers and use a mixer -- keep in
mind, I'm thinking of a radio talk show, presumably with a mixer,
other audio sources, etc. It would look something like this:
+--+--- line out --+---++--+
POTS --| Asterisk || mixer |---| streaming server |
+--+-- line in +---++--+
|| | | |
| CD | | |
SIP Clients, Etc.Mic | Internet
Etc.
Where line out of the Asterisk goes to an input of the mixer
and line in is connected to a monitor port on the mixer.
This would be very simple to do and wouldn't require conferences.
You could map inbound calls to some telephone if you wanted
to screen callers or anything like that and then forward
the call to the console extension when you are ready to
go on the air
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[Asterisk-Users] Terminating VoIP calls with Asterisk
Hi, I'm a newbee to all this asterisk stuff, and after reading a fair amount of docs at the voip-info wiki, I was wondering if it will be possible to work out something as the following scheme: A B +-+ ++ +--+ +-+ | SIP | || Internet |PC| phone line | classic | | | === | router | === | Asterisk | | tel | |Phone| || | FXO card | | | +-+ ++ +--+ +-+ The goal is to be able to pass calls at A's place, being billed as a B call by the telco. Is it possible? If someone has already done it, can you please suggest a good FXO card to do it? I'm specially interested in buying Digium cards, as they seem to have an excellent support, but what would you suggest, a X100P or a TDM400P with a FXO module? I would like to scale that to 4 (maybe 8) lines, and I want a good voice quality on the calls. I've also read on the mailing list, that a factor to get a good quality on the calls is due to both, the FXO card and the line itself (and that you should adjust the gain of card). Do you need some special equipment to do these measures, or can it be done only by software? Also, would you suggest any good book that explains these issues? Thank you very much for any help! Cheers, = Joaquin Cuenca Abela e98cuenc at yahoo dot com __ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
