RE: [Asterisk-Users] Setting Up @Home 0.8 Guide
I have taken all of the text and put it into a message in the forums: http://geekgazette.com/phpbb/viewtopic.php?p=14#14 -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank smith Sent: Wednesday, March 30, 2005 11:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Setting Up @Home 0.8 Guide is there a way you can write those screen shots in to text format on the user guide? I am a blind computer user and am unable to see the examples that are shown on the site. thanks hank - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, March 30, 2005 11:12 PM Subject: [Asterisk-Users] Setting Up @Home 0.8 Guide Because of all of the changes to AMP, we have written up a completely new How-To Guide for [EMAIL PROTECTED] v0.8. Our first example uses BroadVoice for the trunk. http://www.geekgazette.com -Kerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI call fails
Hi!
Can someone help me with a problem I have with CAPI and dialing out
or in? Installed is a B1ISA from AVM.
I have installed chan_capi-0.3.5. In modem.conf I have this entries:
[interfaces]
driver=chan_capi
type=autodetect
dialtype=tone
mode=immediate
msn=144673
device = /dev/ttyI0
device = /dev/ttyI1
in modules.conf:
[modules]
...
load = chan_capi.so
[global]
chan_capi.so=yes
in capi.conf:
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=144673
incomingmsn=*
outgoingmsn=144673
controller=1
softdtmf=1
accountcode=
context=ausgehend
echosquelch=1
echocancel=yes
echotail=64
;callgroup=1
deflect=144673
devices=2
and in extensions.conf:
exten = _3X.,1,Dial,CAPI/144673:${EXTEN:1}
with asterisk -r I get:
Connected to Asterisk 1.0.7 currently running on delta (pid = 1213)
Verbosity is at least 5
delta*CLI capi debug
CAPI Debugging Enabled
-- parse_srv: SRV mapped to host proxy.de.sipgate.net, port 5060
-- Registered SIP 'andreas' at 192.168.1.3 port 5060 expires 1800
-- Saved useragent X-Lite release 1105d for peer andreas
-- Executing Dial(SIP/andreas-7ed4, CAPI/144673:0634187482) in new stack
-- data = 144673:0634187482
-- capi request omsn = 144673
== found capi with omsn = 144673
== CAPI Call CAPI[contr1/144673]/3 -- creating pipe for PLCI=-1
-- Called 144673:0634187482
-- CONNECT_CONF ID=002 #0x0007 LEN=0014
Controller/PLCI/NCCI= 0x101
Info= 0x0
-- CONNECT_CONF ID=002 #0x0007 LEN=0014
Controller/PLCI/NCCI= 0x101
Info= 0x0
== received CONNECT_CONF PLCI = 0x101 INFO = 0
-- DISCONNECT_IND ID=002 #0x0005 LEN=0014
Controller/PLCI/NCCI= 0x101
Reason = 0x3302
== DISCONNECT_IND PLCI=0x101 REASON=0x3302
-- CAPI Hangingup
-- removed pipe for PLCI = 0x101
== No one is available to answer at this time
-- Timeout on SIP/andreas-7ed4
== CDR updated on SIP/andreas-7ed4
-- Executing Hangup(SIP/andreas-7ed4, ) in new stack
== Spawn extension (ausgehend, t, 1) exited non-zero on 'SIP/andreas-7ed4'
X-lite tells me: Call failed: 403 Forbidden
I have no clue what is going wrong.
capi info tells me this:
delta*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
and cat /proc/capi/controllers/1:
name b1isa-340
io 0x340
irq 7
type B1 ISA
ver_driver 3.11-03
ver_cardtype B1
ver_serial 02081722
protocol DSS1
linetype point to multipoint
cardname B1
I tried a lot different settings with no success. Can someone help
me with this? Is the B1 defective or is it the cable?
Thanks in advance!
--
Andreas Meyer
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[Asterisk-Users] Simple authentication
Hi, I would like to create extension, so user will have to enter password, and later he will be prompt for a number to call. My config looks like this (ONLY THE PART OF): exten = 888,1,Ringing(), exten = 888,2,wait(2) exten = 888,3,Background,welcome exten = 888,4,Authenticate(1234|a) exten = 888,5,goto(plan,s,1) [plan] exten = s,1,Playback,pls-entr-num-uwish2-call exten = s,2,wait(10) exten = s,3,Dial(SIP/[EMAIL PROTECTED],60,Ttr) exten = s,4,Hangup The thing is that this is not working the exten = s,3,Dial(SIP/[EMAIL PROTECTED],60,Ttr) calls null number. How do I pass the number to s,3. Can I do a check if caller entered any numbers? Thanks Bart, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to call land line number using wireless land line service through asterisk
dear fellows, i want to make a call from an ip phone to a local pstn number (say 021-699-8256) using a Wireless phone service. asterisk comes in b/w the ip phone and the Wireless phone which is connected to fxs port#1. i just want to dial a number (above mentioned) through ip-phone and asterisk connects a call through Wireless service. the wireless service charge every call i make in this way please reply me if you made a setup of this kind.. i apologiezes to all if these kinds of messages already answered before. if it is then refer me the mail that was answered.. regards. haris Xnet Solutions Inc. Karachi,Paksitan. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
Isamar Maia wrote: Technically speaking not. But Sangoma's support seems to be pretty much better. My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. They're glad to use Asterisk as a selling point for their hardware, but unwilling to donate anything back to the Asterisk community. I'll be glad to stand corrected, but if that assertion is in fact true, we should be careful to do things that actually damage Digium's ability to leverage their development of Asterisk with their hardware sales. I don't understand this *love* for Digium. Digium is a commercial institution, period. If we need to be thankful for Mark Spencer for giving asterisk to the world as many say, I understand and agree. But to protect them specially in my case since I am in Japan and Digium products don't(and it seems that will never) have any support for NTT lines, is kinda no sense. I would better support the Asterisk Fork development that seems to be happening in the underground. BTW, anybody knows their mailing list? I'll be glad to contribute. Isamar Maia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open Source Billing Software
Its the most crappy piece of software i've ever seen. Not to mention that even the installation files for the latest version are incomplete. (try importing the .sql files). Joachim. Kanuri, Seshu (Company IT) wrote: it is by far the most complete billing system. Really? Are you using it now or just pasting here, what you see on the WIKI? I don't think many people here have been using TRABAS as a billing software for Asterisk. Why do you think it is complete? I am a little biased on the opposite as it never worked for me. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max W Blackmer Jr Sent: Wednesday, March 30, 2005 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Open Source Billing Software You Might want to look at Trabas ( http://www.trabas.com/opensource/index.html ) it is by far the most complete billing system. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI call fails
On Thu, 2005-03-31 at 10:01 +0200, Andreas Meyer wrote: REASON=0x3302 This means Protocol error layer 2. Are you able to make outgoing calls any other way using this card? Do you see anything relevant in 'dmesg' when you make outgoing calls, or when incoming calls occur? You don't need to configure modems.conf to use CAPI, btw. -- dwmw2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple authentication
On Thu, 31 Mar 2005 02:11:46 -0600 (CST), Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] wrote: Hi, I would like to create extension, so user will have to enter password, and later he will be prompt for a number to call. Did you look at DISA? This does almost exactly what you're looking for. http://www.voip-info.org/wiki-Asterisk+cmd+DISA Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to call land line number using wireless landline service through asterisk
When you say wireless service to you mean a cellular service or a cordless phone? Why not just have an analog phone line plugged into the asterisk server that can connect to the PSTN to reach the local number? -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Muhammad Haris Sent: Thursday, March 31, 2005 12:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how to call land line number using wireless landline service through asterisk dear fellows, i want to make a call from an ip phone to a local pstn number (say 021-699-8256) using a Wireless phone service. asterisk comes in b/w the ip phone and the Wireless phone which is connected to fxs port#1. i just want to dial a number (above mentioned) through ip-phone and asterisk connects a call through Wireless service. the wireless service charge every call i make in this way please reply me if you made a setup of this kind.. i apologiezes to all if these kinds of messages already answered before. if it is then refer me the mail that was answered.. regards. haris Xnet Solutions Inc. Karachi,Paksitan. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 on Soekris 4801
I love soekris boxes, but in my humble opinion the answer would be be no. Just for yucks set up a 2-3GZ bix and compare it with a 4801, perhaps you will com to a different conclution that I. If some kind sole could port the codecs to use Serin's minipci encryption card, than it might be a different story, Keep in mind that Sarin is hoping on producing two new boards by the end of the year, one with a mobile Athlon-64 --- From the Soekris maillist: Original Message Subject: Re: [Soekris] net5801 net7501 Date: Wed, 23 Mar 2005 15:34:20 -0800 From: Soren Kristensen [EMAIL PROTECTED] Organization: Soekris Engineering To: Jabbar Fagan [EMAIL PROTECTED] CC: [EMAIL PROTECTED] References: [EMAIL PROTECTED] Jabbar Fagan wrote: May I ask why the switch to Intel architecture for the upcoming net5801? Will the net7501 be based on Celeron as well? BTW -When can we expect the new products? Ok, it keep comming up My current product plans: net5501: AMD Geode-2, 500+ Mhz, ddr dimm memory, 4x ethernet, maybe 2x gigabit ethernet option, 2x SATA, 2x PCI slots, 19 1U case, otherwise as rest of family. Target: Late summer 2005, assuming availability of new chips. net7501: AMD Mobile Athlon-64 Mobile Sempron, ddr dimm memory, 2x gigabit ethernet, 2x ethernet, 2x SATA, 2x PCI-X slots, 19 1U case, otherwise as rest of family. Target: Hmm, I need to hire some more people, then maybe late summer 2005 too Best Regards, Soren Kristensen Michael Graves wrote: If I were to try using g.729 over IAX to my prefered ITSPs would a Soekris 4801 be able to handle 3-4 calls at one time? I use Polycom phones running G.711. I'm not certain how much CPU power th transcoding takes. If the phone supports G.729 I suppose I could skip the transcode except for VM. Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 and Asterisk, I think I have a curly one here
This bit me too. Had to turn nat off on the 7960Gs Kristian Kielhofner wrote: Peter J VERNON wrote: Guys.. I have Asterisk CVS-NHEAD-03/19/05-21:56:28 running on a box here and have a couple of Cisco 7960s and a Grandstream phone. I can make calls from the 7960. When I get a call placed to the 7960 the call is setup but there is no audio in either direction. This is for a call placed on the local subnet between extensions so I doubt that it is a NAT issue though I have tried a number of combinations of this to no avail. I have tried firmware versions 6 7 on the Cisco phones, same result. I have tried the phones on two other Asterisk installs and they work fine. I have compared sip.conf with these and can see no differences. If I configure the grandstream up to replace the 7960 it works fine. I have noticed that the src port (TCP port on the phone) increments during the session which seems to be the issue. Anyone seen this before? Any assistance would be appreciated. Regards Peter Peter, This one bit several of us. Upgrade to CVS-Head as of 3/22/2005 or later. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 'RFC3261 transaction matching failed' and 'one-way' communication
Forwarding a call from Asterisk to Microsoft Live Communication Server 2005 via SER (to translate from UDP to TCP), I get a 'one-way' communication (WMessenger user can hear voice but PSTN phone user cannot). Running SER in debug mode, I found: DEBUG: RFC3261 transaction matching failed DEBUG: t_lookup_request: no transaction found Searching Google, I found that this is a known bug in Asterisk (2687): http://bugs.digium.com/bug_view_page.php?bug_id=0002687 I upgraded to 1.0.7 (Debian unstable) but bug is still present. Do I need to compile from CVS? Some other chance? I'm not a compile guru... Thanks in advance Domenico Viggiani ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to call land line number using wireless land line service through asterisk
i dont have anolog phone at my office. anyway i will arrange it for instance. but can u do me a favour? plz resolve y queries... i have connected a anolog phone to Fxs port-1 at asterisk machine now plz send me the configuration of extension.conf to make outbound calls. i had configured zaptel.conf and zapata.conf. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip.conf match
Hello. I have two hired pstn numbers with the same voip provider. I want to distingish in the sip.conf file, what of two phone numbers was dialed, but i don't know how to do the match, because the sip client and the sip host are the same for both numbers. How can i match in sip.conf by the (TO: ) header in sip negotiation? Sorry for my poor english :) Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
Isamar Maia wrote: I don't understand this *love* for Digium. Digium is a commercial institution, period. Yes, but. They are a commercial institution which took an enormous risk by giving away for free what is undeniably their most valuable product. It was a gamble, as it were, of the family jewels. Compare for a moment with Cisco, whose software, as is famously seen written up on this very list, nickels and dimes its customers to death. But to protect them specially in my case since I am in Japan and Digium products don't(and it seems that will never) have any support for NTT lines, is kinda no sense. The kinda sense of it is that many of us believe that we are furthering the cause of Asterisk development by indeed giving preferential treatment to Digium, in those cases where it can be done economically. It's an investment in the future of Asterisk. It's called voting with your pocketbook. I would better support the Asterisk Fork development that seems to be happening in the underground. BTW, anybody knows their mailing list? I'll be glad to contribute. I do know the address of one such list, and I monitor it assiduously, reading every message because my interest in Asterisk is pretty absolute. If I recall correctly, the last mail on that list was either the third or fourth mail that was sent, a couple of days after the list was established, back a couple of months. As far as I know nothing has happened, at all, since then. Forks are cheap. Talking about forks is even cheaper. Forks that appear to be actual improvements over the current Asterisk codebase--nothwithstanding the criticism it receives--appear to be, for now, a null set. I'm sorry you have trouble understanding this. I feel that for many of us it is pretty clear. Thanks. b. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk -- PABX
--- [EMAIL PROTECTED] wrote: At the moment all I know is that they have Siemens PBX system. They will give me more details soon. since HiPath4x00 V1.0 you can use oh323 and HG3550 (STMI board in the HiPath) for interconnection between Siemens HiPath4x00 PBX and Asterisk. domé __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
Isamar Maia wrote: I don't understand this *love* for Digium. Digium is a commercial institution, period. Yes, but. They are a commercial institution which took an enormous risk by giving away for free what is undeniably their most valuable product. So, if Linus Torvalds had a company I would need to buy products from him? If they assumed this risk, great! I will remember to send a postcard in the Christmas to them. More hardware companies support Asterisk with Zap drivers, cheaper will be the boards, better quality will be provided and in the end of the day, the community will have all the benefits. The name of it is competition. Or it's a monopoly? Maybe Japan or other countries with own crazy standards are not a commercial interest of Digium like they are for Avaya, Dialogic, Aculab and stuff... the open and free competition should happen because the world is not USA and AFAIK it's GPL. I'm sorry you have trouble understanding this. I feel that for many of us it is pretty clear. Yes. I see. Very clear. Isamar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF detection in dial macro
Hi all,
Has anyone got the call screening sample to pickup DTMF correctly ? I
have tried with the latest HEAD release and the dial macro gets executed
all the way up until the Read command where it sits until the timeout is
triggered no matter what DTMF tones you send it. Asterisk responds with
User entered nothing.. I have tried this on a variety of extensions
with the same result although a simple DTMF test directly on the ingress
leg of the call works perfectly...
The config I am using is as follows:
exten = 123,1,Wait(0.2)
exten = 123,2,Playback(screen-record)
exten = 123,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten = 123,4,Record(${SCREEN_FILE}.gsm|6|25)
exten = 123,5,Dial(Zap/g1/01276459906|60|gM(screen^${SCREEN_FILE}))
exten = 123,6,Voicemail([EMAIL PROTECTED])
[macro-screen]
exten = s,1,Wait(0.2)
exten = s,2,Playback(screen-from)
exten = s,3,Playback(${ARG1})
exten = s,4,Read(ACCEPT|screen-accept|1)
exten = s,5,GotoIf($[${ACCEPT} = 1 ] ?7:6)
exten = s,6,SetVar(MACRO_RESULT=CONTINUE)
exten = s,7,System(/bin/rm ${ARG1})
I am guessing this is a bug of sorts relating to the detection code not being
applied to the egress
channel but I dont know enough about how this works yet to debug it...
Any help greatfully received !
Tristan.
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[Asterisk-Users] Music Answer while waiting
Hi, If I want a user to, while waiting for a transfer after responding to an IVR, to listen to music instead of a ring sound, what is the change should i do in extensions.conf? Is it on the IVR menu or on the optional extension Txs, Robson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Australia and SetCallerID
On Thu, Mar 31, 2005 at 01:58:21PM +1130, Craig wrote: From my investigations, I can't find any carrier in Au that does allow it to be set outside the allocated range, can't even get one to set it to our 1300 number, have been told the ACA doesn't permit it in Au, but not certain on that. Correct. It's bordering on illegal to lie about your callerID (not quite though, because it's not legislated). Certainly you're not going to be able to do it on any basic commercial service. Carrier interconnects can do it obviously. The reason is related to emergency services (amongst other things), a 1300 number has no physical location. Origination numbers can be geographic or mobile but must be associated with a specific location or device. Technically, if you have a 100 number range that you split over multiple locations, you're required to indicate which numbers are where. Doesn't happen often though. It even goes so far as if you're transiting a call you must preserve the originating number. This means if you setup a calling card platform as a carrier you may have a few extra requirements. My understanding many pri in the US can be set to almost any number. So I've heard, can't quite understand the benefits, but this is the way things are. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] File permissions and ownership
On Wed, Mar 30, 2005 at 11:36:18AM -0800, Kenneth Porter wrote: Excellent, thanks for the info! I was mostly worried about opening privileged ports, but an initial test showed only high ports opened. I'd guess that only files asterisk needs to write need to be owned by the asterisk user, and the other files (eg. sounds) can be owned by root and made world-readable. The only issue is that as non-root, asterisk can't set the TOS bits (though iptables can do it for you) and it can't set its own priority (though nice/renice can help there). -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
On Thu, Mar 31, 2005 at 02:35:07AM -0500, Kris Edwards wrote: Well, I'm certainly not selling xten.. Perhaps my enthusiasm extends from my disgust with everything else. In particular, kphone, and sjphone. I have noticed latency with xten in meetme, but if I just dial somebody it works better than anything I've tried (so far.. I've only spend about 1 hour talktime). Anyway, I'm certainly more hip on open source, and can't wait to try gnomemeetings sip once I can actually get it to compile :/ I have to agree though, I tried a lot of softphones under linux and the xten was the first one that worked. Not just that, it worked *perfectly* first time, no whacky obscure problems. Now if only Firefly worked under linux, that's be really cool... -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] External line hangup
Hi, I have discovered something strange with my asterisk box. After having figured out how to get incoming calls to work on my TDM22B (TDM400P) card, i am having an issue in that when the outside caller hangs up the phone before the phone is answered on extention end, the extention rings till voicemail or till deadline and doesn't detect the hangup. Is there something i might have missed in the configuration files? Please let me know where i could look. thanks Sascha ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Zaptel Periodic Reset
In article [EMAIL PROTECTED], Rod Bacon [EMAIL PROTECTED] wrote: I have noticed whilst being connected to the console of my * server, that my PRI interface (Digium TE410P) periodically reinitialises itself. This server is currently not actively used, and each time the reset happens the card is idle. Is this normal behaviour, or does it signify a problem? Normal behaviour. Once an hour, all idle channels get restarted, but active channels are left undisturbed. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reject second IAX call
Hi, is there a configuration in iax.conf to specify that if a call goes to that peer, a second call should not be allowed. Specifically, I do this: Dial(IAX2/iaxcomm) # in extensions.conf for a specific extension in iax.conf: [iaxcomm] type=friend mailbox=20 accountcode=iaxcomm username=iaxcomm host=dynamic auth=md5,plaintext,rsa secret=fksjdfh73 ; changed context=local-iaxcomm permit=192.168.10.0/24 allow=ulaw is there an option to disable a 2nd call? thank you. PS: the real problem in my case is that for some reason IAXcomm sees a second call coming in after 30 sec - 1 minute on 2 over 10 incoming calls. This phantom call must be disconnected to resume the real call. Funny duh? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music Answer while waiting
On Mar 31, 2005 1:00 PM, Robson Ribeiro [EMAIL PROTECTED] wrote: Hi, If I want a user to, while waiting for a transfer after responding to an IVR, to listen to music instead of a ring sound, what is the change should i do in extensions.conf? Is it on the IVR menu or on the optional extension The change id one in the dial command that calls the extension show application dial in the cli will help look at the m option ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reject second IAX call
On Mar 31, 2005 12:31 PM, Marc SCHAEFER [EMAIL PROTECTED] wrote: Hi, is there a configuration in iax.conf to specify that if a call goes to that peer, a second call should not be allowed. Specifically, I do this: Dial(IAX2/iaxcomm) # in extensions.conf for a specific extension in iax.conf: [iaxcomm] type=friend mailbox=20 accountcode=iaxcomm username=iaxcomm host=dynamic auth=md5,plaintext,rsa secret=fksjdfh73 ; changed context=local-iaxcomm permit=192.168.10.0/24 allow=ulaw is there an option to disable a 2nd call? thank you. look on wiki for set group and check group this can do what you need ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] early B3 connect with TE110P
hello from germany,
i'm using a TE110P in E1-mode with asterisk as a VOIPPSTN gateway. i can
dial out with the sip-phones and everything is ok, but when i dial a wrong
phonenumber, with a normal phone i will hear a message telling that, but
asterisk passes no audio to the phone, like it worked with chan_capi and
isdn with early B3 connect enabled. the best would be to do all the
status-signalling like busytones etc. via audio like its done in grandmas
phone. does anyone know how to achieve this with zaptel?
thanks in advance!
dave
p.s.: my configfiles:
/etc/zaptel.conf:
--snip--
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone = nl
defaultzone=nl
--snap--
zapata.conf:
--snip--
[channels]
switchtype=euroisdn
pridialplan=local
prilocaldialplan=local
internationalprefix=00
nationalprefix=0
usecallingpres=no
busydetect=no ; not need on pri
;callprogress=yes ; was yes but wiki says experimatley could be
produce ha
ngups
callwaitingcallerid=yes ; show callerid on callwaitingcalls
echotraining=no
echocancel=no
echocancelwhenbridged=no
overlapdial=no
;immediate=yes
;callerid=asreceived
callerid=no
language=de
rxgain=0.0
txgain=0.0
group=1
signalling=pri_cpe
context=default
channel = 1-15,17-31
--snap--
extensions.conf:
--snip--
[general]
static=yes
writeprotect=no
autofallthrough=yes
[default]
;;outbound via TE110P
exten = _0.,1,Dial(Zap/g1/${EXTEN},30)
exten = _0.,102,Busy
--snap--
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[Asterisk-Users] We require Asterisk configuration and support consultants
We require Asterisk configuration and support consultants to concentrate on the "other" business issues. The expected functions of the consultant(s) are: * Help in configuring setting up complete Asterisk system (h/w setup and LINUX setup is handled by us). System may include Digium PRI, ZapTel, H323, SIP components. * Corresponding party on our side is an expert programmer (proficient in C)/system administrator (proficient in LINUX). * Allcommunicationshould usee-mail and/or voice/fax. The language is English. Any offer for the above listed role is to be communicated to private mail Tayfun YIGIT [EMAIL PROTECTED] The offer may include "per incident" fee and/or weekly rate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] setting SIP to dial PSTN with TDM400P
I've setup * with TDM400P w/1 FXS, 1 FXO modules. I've one analog phone connected to TDM400P FXS module, 1 PSTN line to one of the FXO module(ZAP) , and IP phone connected to asterisk on LAN. The calls between SIPs and zap phone (fxs) are OK. But 2 issues cannot be solved: 1. To dial to PSTN via zap phone, the setup in extensions.conf with the following exten = _Nxx, 1, zap/1 doesn't work. Does anyone can give me suggestion that what did I do wrong to make the setting: 2. When trying using SIP phone to dial PSTN, I got no luck. Please advice if any solution. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems editing oh323 configuration parameters
Checking the oh323 configuration on asterisk console gives the following result below. I'm editing the /etc/asterisk/oh323.conf file to correct the parameters, but the result doesn't change. I didn't receive any error massages during the installation of asterisk-oh323-0.7.1 channel driver. So what might be wrong? localhost*CLI oh323 show conflocalhost*CLIConfiguration of OpenH323 channel driver--Version: 0.7.1Listening on address: :1720Gatekeeper used: FailedFastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFFSupported formats in pref. order:Jitter buffer limits (min/max): 20-100 msTCP port range: 5000 - 31000UDP (RAS) port range: 5000 - 31000UDP (RTP) port range: 1 - 2IP Type-of-Service value: 0User input mode: 2Max number of inbound H.323 calls: 0Max number of outbound H.323 calls: 0Max number of simultaneous H.323 calls: -1Max call rate (ingress direction): 99.00/30 Thanks in advance for any help, Cenk Yabas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Installing asterisk and components
In which directories I should install asterisk, chan_capi, and modem driver? And did I forgot something to get asterisk functional? what is best way to test quick is the pbx working, at this point I only have HFC card for external isdn lines? I have RH9 so Linux kernel should be fine? Thank you for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems editing oh323 configuration parameters
Did you copy the oh323.conf file from the asterisk-oh323 package ? Could you show what it looks like ? Are the file permissions ok ? Yves Cenk Yabas wrote: Checking the oh323 configuration on asterisk console gives the following result below. I'm editing the /etc/asterisk/oh323.conf file to correct the parameters, but the result doesn't change. I didn't receive any error massages during the installation of asterisk-oh323-0.7.1 channel driver. So what might be wrong? localhost*CLI oh323 show conf localhost*CLI Configuration of OpenH323 channel driver -- Version: 0.7.1 Listening on address: :1720 Gatekeeper used: Failed FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF Supported formats in pref. order: Jitter buffer limits (min/max): 20-100 ms TCP port range: 5000 - 31000 UDP (RAS) port range: 5000 - 31000 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 0 User input mode: 2 Max number of inbound H.323 calls: 0 Max number of outbound H.323 calls: 0 Max number of simultaneous H.323 calls: -1 Max call rate (ingress direction): 99.00/30 Thanks in advance for any help, Cenk Yabas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems editing oh323 configuration parameters
You dont have any codecs configured in your oh323 conf. also FastStart with H245 tunneling should be enabled to get the best call-setup out of h323. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cenk Yabas Sent: Thursday, March 31, 2005 7:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problems editing oh323 configuration parameters Checking the oh323 configuration on asterisk console gives the following result below. I'm editing the /etc/asterisk/oh323.conf file to correct the parameters, but the result doesn't change. I didn't receive any error massages during the installation of asterisk-oh323-0.7.1 channel driver. So what might be wrong? localhost*CLI oh323 show conf localhost*CLI Configuration of OpenH323 channel driver -- Version: 0.7.1 Listening on address: :1720 Gatekeeper used: Failed FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF Supported formats in pref. order: Jitter buffer limits (min/max): 20-100 ms TCP port range: 5000 - 31000 UDP (RAS) port range: 5000 - 31000 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 0 User input mode: 2 Max number of inbound H.323 calls: 0 Max number of outbound H.323 calls: 0 Max number of simultaneous H.323 calls: -1 Max call rate (ingress direction): 99.00/30 Thanks in advance for any help, Cenk Yabas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Installing asterisk and components
Checkout http://www.voip-wiki.org as it relates to asterisk. There are a number of useful guides on how to setup and run asterisk. Btw, all the config files should be located in /etc/asterisk. RH9 should be fine to run asterisk. Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, March 31, 2005 7:29 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Installing asterisk and components In which directories I should install asterisk, chan_capi, and modem driver? And did I forgot something to get asterisk functional? what is best way to test quick is the pbx working, at this point I only have HFC card for external isdn lines? I have RH9 so Linux kernel should be fine? Thank you for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cmd Authenticiation
Simon, I am not sure if I understand you question properly. However, you can configure password for each user (peer or friend) in corresponding channel configuration file (i.e. sip.conf) HTH Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon Sent: Wednesday, March 30, 2005 10:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cmd Authenticiation Hi folks, Sorry to post a simple command, I am deep into this and hope any help from the experts. I am using the command Authenticate as explained in wi-ki:I am managed to authenticiate with a single global passwordbut my requirement will every user have their own password and contexts to callPlease help meThank youSimon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ACD queue question
After I changed from leastrecent I did reload asterisk and waited about an hour and nothing changed. So I restarted asterisk and waited another hour, but it was still calling the agents in the order that they are listed in the agents.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Umar Sear Sent: Thursday, March 31, 2005 1:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ACD queue question are you restarting asterisk or reloading after changing you configuration. Umar On Wed, 30 Mar 2005 19:33:42 -0600, Eric Rees [EMAIL PROTECTED] wrote: I tried leastrecent. I did change the strategy, but didn't make a difference. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Wednesday, March 30, 2005 6:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] ACD queue question Using which strategy? Remember, if you change strategies and reload, it'll forget where it was and start over. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees Sent: Wednesday, March 30, 2005 6:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ACD queue question That's what I thought would happen, but after about an hour and 100 or so incoming calls, it was still ringing the agents in the order that they were listed in the agents.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Tuesday, March 29, 2005 10:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] ACD queue question The first call for each agent probably goes that way, but then after a few calls have rolled through the queue, the strategy you specify (like LeastRecent) should come into play. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees Sent: Tuesday, March 29, 2005 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ACD queue question I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
True. I think Digium's USA bias is clearly demonstrated by their lack of a BRI ISDN product. Most of the rest of the world use it in abudnace yet Digium do not see fit to service this market because it is not big in the US. very poor... On Thu, 31 Mar 2005 18:32:40 +0900 (JST), Isamar Maia [EMAIL PROTECTED] wrote: Isamar Maia wrote: I don't understand this *love* for Digium. Digium is a commercial institution, period. Yes, but. They are a commercial institution which took an enormous risk by giving away for free what is undeniably their most valuable product. So, if Linus Torvalds had a company I would need to buy products from him? If they assumed this risk, great! I will remember to send a postcard in the Christmas to them. More hardware companies support Asterisk with Zap drivers, cheaper will be the boards, better quality will be provided and in the end of the day, the community will have all the benefits. The name of it is competition. Or it's a monopoly? Maybe Japan or other countries with own crazy standards are not a commercial interest of Digium like they are for Avaya, Dialogic, Aculab and stuff... the open and free competition should happen because the world is not USA and AFAIK it's GPL. I'm sorry you have trouble understanding this. I feel that for many of us it is pretty clear. Yes. I see. Very clear. Isamar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma VS. Digium
Hello, I need to correct myself on one of the points I made in my reply last night. As a very polite developer from Sangoma stated to me(with evidence I might add)they have in the past and continue to today contribute code to GPL Asterisk. It doesn't say so on their website but their developers have been bug-checking, patching and contributing new code to Asterisk for some time now. They just started directly giving credit from Sangoma for some of these contributions in the bugtracker starting this week. While it is true that they probably don't have as many full-time dedicated Asterisk developers as Digium does, a portion of a Sangoma AFT card purchase will go towards further development of Asterisk. So you can feel a little less-bad about buying those Sangoma cards now. MATT--- -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Thursday, March 31, 2005 2:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sangoma VS. Digium Isamar Maia wrote: Technically speaking not. But Sangoma's support seems to be pretty much better. My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. They're glad to use Asterisk as a selling point for their hardware, but unwilling to donate anything back to the Asterisk community. I'll be glad to stand corrected, but if that assertion is in fact true, we should be careful to do things that actually damage Digium's ability to leverage their development of Asterisk with their hardware sales. FWIW. b. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CAPI call fails
Hi! David Woodhouse [EMAIL PROTECTED] wrote: On Thu, 2005-03-31 at 10:01 +0200, Andreas Meyer wrote: REASON=0x3302 This means Protocol error layer 2. Are you able to make outgoing calls any other way using this card? Do you see anything relevant in 'dmesg' when you make outgoing calls, or when incoming calls occur? You don't need to configure modems.conf to use CAPI, btw. ah, thanks! I have this output after the machine rebooted: ... Adding Swap: 511992k swap-space (priority 42) CAPI-driver Rev 1.1.4.1: loaded capifs: Rev 1.1.4.1 capi20: started up with major 68 kcapi: capi20 attached capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs) CSLIP: code copyright 1989 Regents of the University of California ISDN subsystem Rev: 1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1 loaded kcapi: capidrv attached kcapi: appl 1 up capidrv: Rev 1.1.4.1: loaded b1: revision 1.1.4.1 b1isa: revision 1.1.4.1 kcapi: driver b1isa attached kcapi: Controller 1: b1isa-340 attached b1isa: AVM B1 ISA at i/o 0x340, irq 7, revision 255 b1isa-340: card 1 B1 ready. b1isa-340: card 1 Protocol: DSS1 b1isa-340: card 1 Linetype: point to multipoint b1isa-340: B1-card (3.11-03) now active kcapi: card 1 b1isa-340 ready. kcapi: notify up contr 1 capidrv: controller 1 up capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capi: controller 1 up via-rhine.c:v1.10-LK1.1.19 July-12-2003 Written by Donald Becker http://www.scyld.com/network/via-rhine.html PCI: Found IRQ 11 for device 00:11.0 PCI: Sharing IRQ 11 with 00:07.2 eth0: VIA VT6102 Rhine-II at 0xec00, 00:05:5d:a3:56:90, IRQ 11. eth0: MII PHY found at address 8, status 0x782d advertising 01e1 Link 0021. ne2k-pci.c:v1.02 10/19/2000 D. Becker/P. Gortmaker http://www.scyld.com/network/ne2k-pci.html PCI: Found IRQ 12 for device 00:0f.0 eth1: RealTek RTL-8029 found at 0xe400, IRQ 12, 00:00:B4:9C:51:15. usb.c: registered new driver usbdevfs usb.c: registered new driver hub usb-uhci.c: $Revision: 1.275 $ time 13:14:03 Feb 15 2005 usb-uhci.c: High bandwidth mode enabled PCI: Found IRQ 11 for device 00:07.2 PCI: Sharing IRQ 11 with 00:11.0 usb-uhci.c: USB UHCI at I/O 0xe000, IRQ 11 usb-uhci.c: Detected 2 ports usb.c: new USB bus registered, assigned bus number 1 hub.c: USB hub found hub.c: 2 ports detected usb-uhci.c: v1.275:USB Universal Host Controller Interface driver IPv6 v0.8 for NET4.0 IPv6 over IPv4 tunneling driver eth0: Promiscuous mode enabled. device eth0 entered promiscuous mode eth0: no IPv6 routers present eth1: no IPv6 routers present eth0: Promiscuous mode enabled. kcapi: appl 2 up kcapi: appl 2 releasing(1) kcapi: appl 2 down kcapi: appl 2 up capidrv-1: DISCONNECT_IND reason 0x3301 (Protocol error layer 1 (broken line or B-channel removed by signalling protocol)) for plci 0x101 capidrv-1: DISCONNECT_IND reason 0x3302 (Protocol error layer 2) for plci 0x101 I don't know where to start. Dialing out gives no messages in the logfile. Dialing in on this number gives busy on the phone (analog- or ISDN-phone) and no messages in the logfile. If I could get another ISDN-card running with CAPI and SuSE I would try another card, but the B1 ist the only one I got. I can't get a FritzClassic to work with CAPI on this SuSE-Server. I also tried sending out a SMS with yaps using the B1. I get: delta:/var/log # yaps 01757052847 hi Found service D1 for 01757052847 Sending following message: 01757052847 (D1, 01757052847): hi (sent by A.Meyer!) Trying to open /dev/ttyI0 for modem standard [Hangup] [Send] cr [Cmd Mdzz 200] [Send] ATZcr [Expect] crATZcrcrlfOK got OK [Send] ATE144674cr [Expect] crlfATE144674crcrlfOK got OK Using modem standard at 38400 bps, 8n1 over /dev/ttyI0 Trying do dial 01712521001 [Send] ATD01712521001cr [Expect] crlfATD01712521001crcrlfBUSY got BUSY Unable to dial D1 Thank you! -- Andreas Meyer ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma VS. Digium
What about pricing of the Sangoma compared to Digium, is it comparable? Can Sangoma card handle modem data incoming calls at all? Selon mattf [EMAIL PROTECTED]: Hello, I need to correct myself on one of the points I made in my reply last night. As a very polite developer from Sangoma stated to me(with evidence I might add)they have in the past and continue to today contribute code to GPL Asterisk. It doesn't say so on their website but their developers have been bug-checking, patching and contributing new code to Asterisk for some time now. They just started directly giving credit from Sangoma for some of these contributions in the bugtracker starting this week. While it is true that they probably don't have as many full-time dedicated Asterisk developers as Digium does, a portion of a Sangoma AFT card purchase will go towards further development of Asterisk. So you can feel a little less-bad about buying those Sangoma cards now. MATT--- -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Thursday, March 31, 2005 2:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sangoma VS. Digium Isamar Maia wrote: Technically speaking not. But Sangoma's support seems to be pretty much better. My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. They're glad to use Asterisk as a selling point for their hardware, but unwilling to donate anything back to the Asterisk community. I'll be glad to stand corrected, but if that assertion is in fact true, we should be careful to do things that actually damage Digium's ability to leverage their development of Asterisk with their hardware sales. FWIW. b. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sms and DDI UK
Has anyone had any luck with being able to register a DDI with SMS in the UK ? If so, how have you done it ? Anything I send to 0 with a reset message gets sent back to my main ISDN number, even though I have specified a callerid within my DDI range. If I use the same code to dial a number instead of sms, then the callerid is the same as the DDI, so I know that the SetCIDDNum is changing the callerid. Julian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cvs-head from 3/31/05 fails to load
Cross posted on purpose FYI, just upgraded from cvs-head from March 23 to this morning (March 31). All compiles and installs completed normal. Loading asterisk via safe_asterisk (or asterisk -cdvvv) fails with the standard oche... message. Piped the output to a text file and it appears the cdr_custom.so is failing to load. Installed a noload for that module and asterisk now loads properly. Looks like someone needs to do a little more work with that module. Also noticed in asterisk/configs that cdr_custom.conf is not called cdr_custom.conf.sample like the others. FYI. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ser - asterisk -cisco gateway
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
hi,
we have the ser sip-proxy for registration and we forwarding
the call to our cisco gateway and it works.
but now we will forwarding the calls to the asterisk and
the asterisk shoud forward the calls to our gw (via sip not h323).
how must i configure the asterisk
ser.cfg
if(uri =~sip:1024#){
~ log(1,Forwarding to Asterisk\n);
~ setflag(1);
~ rewritehostport(192.168.1.3:5061);
~ t_relay();
}
asterisk
thanks hans
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Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFCS/e6ouYj3oyEw4wRAoseAKCffEjSqxRGPmZaJYawqdoFrVjURACdHIXt
98DkG/axeJ4Gp6ENnMd0shk=
=ik0/
-END PGP SIGNATURE-
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[Asterisk-Users] sharing asterisk among several companies
Hello, I am trying to configure Asterisk to be used by two (or more) different remote companies, sharing the same instance of Asterisk on my host. By setting specific entry contexts for each sip user, I can repeat extensions among companies. My question is: is it possible to have repeated users on sip.conf being identified by their different passwords? I tried to do that but got an authentication failure. Is there a way to do this? Or I should always have different usernames? Thank you Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
On Thursday 31 March 2005 02:43, Brian Capouch wrote: Isamar Maia wrote: Technically speaking not. But Sangoma's support seems to be pretty much better. My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. They're glad to use Asterisk as a selling point for their hardware, but unwilling to donate anything back to the Asterisk community. It really does become an interesting debate. Do you lower your own ability to survive by using a lower quality product/service, to help ensure that the main product continues. Or do you help the main product survive by putting yourself at risk? For better or worse we are all also the effect of Digium's policies and decisions. Not to say that they have not done an outstanding job getting Asterisk to what it is. Likewise Digium is at the effect of what the community does. In the long run one needs to find a balance where everyone can win. Usually that is done by plain market influence. If they don't buy it, you won't be able to make it for very long. Indeed we all would be poorer if Digium could not continue the work. But so too, do they need to ensure that they are staying close to community needs, while making sure they DO make the right decisions. I think it's fair to say that Digium is more right than wrong, as their course have taken them this far. One does however need to reevaluate positions and directions every now and then, and be willing to change course should it so require. I'll be glad to stand corrected, but if that assertion is in fact true, we should be careful to do things that actually damage Digium's ability to leverage their development of Asterisk with their hardware sales. My view of Asterisk has made me put my money where my mouth is by betting the farm on Asterisk. I have put everything I have into a position of making a living with Asterisk, so I too depend on it to survive. But in the end I have to ensure that my decisions keep food on our table. Whether I choose Sangoma or Digium cards will be based on what I perceive to be the most long term survival thing to do. Of course, if I end up making a good business out of Sangoma and Asterisk, nothing will stop me from paying license fee's to Digium, which will be more profitable than selling me a card anyway. So I see that Digium should be making enough money from all of us, each contributing in a different way. In fact at this point Asterisk is poised to become a major influence in the market as people world wide is waking up to it's potential. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
On March 30, 2005 10:26 pm, Kristian Kielhofner wrote: It is obvious that Asterisk/TDM support from Sangoma is (and has been) secondary. Their cards support data like no other. Excellent. Voice, on the other hand, appears to be immature. I respectfully disagree. Sangoma's voice capabilities are no less and no more mature than Digium's voice capabilities. I use cards from both Sangoma and Digium. Both seem to work well but (and it does pain me to say it, it really does) Digium's cards seem FAR more finicky about the type of hardware they'll run reliably on. Sangoma's cards you can pretty much throw into any system and they work. Shared interrupts and oddball PCI chipsets included. I do believe, however, that this is merely a driver issue. If I were a more competent driver programmer I would certainly dive into this headfirst. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing asterisk among several companies
--On Thursday, March 31, 2005 10:20 AM -0300 Dov Bigio [EMAIL PROTECTED] wrote: My question is: is it possible to have repeated users on sip.conf being identified by their different passwords? I tried to do that but got an authentication failure. Is there a way to do this? Or I should always have different usernames? What about the case where a user works at two or more companies? For instance, an employee at one could have a permanent consultant desk at the other. Can one call him at home using either company's extension, and can he identify which identity is being contacted? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] cvs-head from 3/31/05 fails to load
Rich Adamson wrote: Cross posted on purpose FYI, just upgraded from cvs-head from March 23 to this morning (March 31). All compiles and installs completed normal. Loading asterisk via safe_asterisk (or asterisk -cdvvv) fails with the standard oche... message. Piped the output to a text file and it appears the cdr_custom.so is failing to load. Installed a noload for that module and asterisk now loads properly. Looks like someone needs to do a little more work with that module. There's a patch in the bug tracker since a few hours. Always check there :-) Also noticed in asterisk/configs that cdr_custom.conf is not called cdr_custom.conf.sample like the others. FYI. That needs to be fixed. Thank you! /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP not working in GUI
I have recently installed Asterisk @ 8.0 and loading it fine and setup the ip addressing and change the default password. But when I access the gui from a computer on the network I can pull up the gui but the amp link doesn't work. http://192.168.1.x/admin doesn't work any leads on this. Regards, --- Otis Surratt Jr. / [EMAIL PROTECTED] --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sharing asterisk among several companies
My question is: is it possible to have repeated users on sip.conf being identified by their different passwords? I tried to do that but got an authentication failure. Is there a way to do this? Or I should always have different usernames? I think it would be less crazy best if you had a naming scheme, such as companyname-username, and stuck with it. Cheers, Jean-Michel. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automatic Configuration Tools?
March 31, 2005 Hello Asterisk-users: I am interested in automatic configurations based on typical legacy phone systems configurations. In other words, after installing Asterisk- I don't want to have to learn the mechanics of the PBX, I just want it to work out of the box. Given that most PBX solutions follow similar configurations, a selection of say 5 or 6 configurations could be selectable in a menu. Obviously some people will need a fully customized solution, and that's fine- I am just looking to reduce my deployment costs. Any ideas? Thanks. Mark Watson Internet Consultant Mark Watson Consulting MWCNETWORK.COM 421 Pittsfield Drive Worthington, OH 43085 (614) 923-3929 VOIP Phone (614) 589-2225 Cell [EMAIL PROTECTED] http://www.mwcnetwork.com inline: title.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Time sync on PRI
Hi! I have this setup at a customer: PRI - (port 1) TE410P (port 2) - PABC | Asterisk Before the Asterisk part was inserted the customer claims that their PABC automatic changed the clock acourding to daylight saving time from the PRI. Now the customer says that it is not working any more. We are using pri_net signalling up against the PABC and pri_cpe on the other interface. Does Asterisk send time syncronisation on pri_net signalling? Is there a configuration setting that enables that? Is it possible to sync the computer clock with the time from the PRI? The Asterisk server is not connected to a network so NTP is not an option. -- Morten Isaksen http://www.aub.dk/~misak/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
On Thu, 31 Mar 2005, Eric Bishop wrote: True. I think Digium's USA bias is clearly demonstrated by their lack of a BRI ISDN product. Most of the rest of the world use it in abudnace yet Digium do not see fit to service this market because it is not big in the US. very poor... Why on earth would Digium develop a ISDN BRI card while you can buy a HFC-S card that will work anywhere in Europe for less than 30 dollars? (The quad bri cards are overpriced, EUR 600 is just too much for such a simple card). That would be a waste of time and money better spent on other development. If you use bristuff and use the florz patch you have a very stable and solid product. (Bristuff without florz patch sucks, I really don't understand why the GPL'ed florz patch isn't included in bristuff by default). It would be nice if Digium would accept the bristuff patch at some stage and include it in asterisk. Regarding Sangoma vs. Digium cards, clearly Digium is providing a product at a very reasonable price (both the T1/E1 cards as well as * itself) and I really do not see any reason to not support them by buying Sangoma. Just my $0.02 :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom220
Good day all I'm looking for someone with good knowledge of the way the snom220 transfer I want to know how to do a consultative transfer on the second call I.o.w if a call come in,A and another call come in B and B asks to be transfered to exten 200,I want to speak to 200 1st and the transfer B to 200. Please Help Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Many analog lines
Hi, how to use Asterisk where I need to have lets say 40 analog lines. Any ideas? Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk as Cisco Call-Manager - dial out to PSTN
Hi Maron, Thank you for your answer! I use a simple cisco router 2621XM as call manager with the following configuration: interface Loopback79 description ALT-VoIP-Gateway ip address 10.xxx 255.255.255.255 h323-gateway voip interface h323-gateway voip id Ldnxxx ipaddr 10.xxx 1719 priority 120 h323-gateway voip h323-id [EMAIL PROTECTED] h323-gateway voip tech-prefix 301 h323-gateway voip bind srcaddr 10.xxx The structure is Sip-phone SIP Asterisk as call-manager (extension 399) H.323 cisco gatekeeper (extension ) H.323 cisco call-manager (extension 302) E1 PSTN Iif I dial now with the Sip-phone: 302 [PSTN number (handy number, .)] I should be able to telephone the the PSTN of the call manager with the extension 302. It works within cisco devices perfectly but not with asterisk. Can you tell me your experiences and practices?? Thanks a lot!! Mario Hi Mario.What kind of Cisco gateway are you using, I swapped an Cisco Call Manager 4.0 for Asterisk, and am using 12 gateways worldwide for PSTN access. However using SIP, which the gateways (Call Manager Express on 1760 routers) support very well for trunking.I've found that H323 is even buggy between the CME gateways from Cisco.Regards,Maron KristoferssonMario Spendier wrote: Hi all,Im running Asterisk since two days, and its really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me? Im running against closed doors in this problem!!! If I phone over a Cisco call manager it works, so the failure is Asterisk based.-- Executing NoOp(SIP/12345-454d, call for ) in new stack -- Executing Dial(SIP/12345-454d, OH323/ ) in new stack -- H.323 call to with codec alaw -- Called -- H.323 call 'ip$localhost/27230' cleared, reason 24 (Call ended with Q.931 cause) -- Hungup 'OH323/L27230'Thanks a lot!!!Mario ___ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ser, asterisk and conferencing
Hi List, Can I use asterisk to enable call conferencing? I'm using ser for the UA's to register, can I do something like if they dial a certain digits, it will forward it asterisk and use asterisks meetme feature? can i do meetme using only sip? Sorry for my terms, hope you understand my question. Regards, Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Concurrent Call in Asterisk
Hi All, Is it possible to have only one SIP account that is shared by several users ? I am currently setting up one asterisk box for a small company (around 7 users). Can all of them make simultaneous call using only one SIP account for termination or I have to setup individual account for all of them (which will be very troublesome on my side as I have to keep reminding them to top up , would be good if I just manage one account) ? Thanks in advance. Stephen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sms and DDI UK
On Thu, 2005-03-31 at 14:10 +0100, Asterisk wrote: Has anyone had any luck with being able to register a DDI with SMS in the UK ? If so, how have you done it ? Anything I send to 0 with a reset message gets sent back to my main ISDN number, even though I have specified a callerid within my DDI range. I did it using 'smsq --queue 586671 0 register' iirc. Certainly I've had both incoming and outgoing calls working on more than just the network number. At the moment they're _all_ on a DDI number, because I've switched to using CAPI and it's using that number for all outgoing calls. -- dwmw2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP not working in GUI
[EMAIL PROTECTED] wrote: I have recently installed Asterisk @ 8.0 and loading it fine and setup the ip addressing and change the default password. But when I access the gui from a computer on the network I can pull up the gui but the amp link doesn't work. http://192.168.1.x/admin doesn't work any leads on this. Regards, --- Otis Surratt Jr. / [EMAIL PROTECTED] --- http://lists.digium.com/mailman/listinfo/asterisk-users You might try the source forge forum for [EMAIL PROTECTED] http://sourceforge.net/forum/forum.php?forum_id=420324 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automatic Configuration Tools?
You could easily write some perl scripts that create you the right config files depending on selectable configurations ... Just an idea. Yves Mark R. Watson wrote: MWCNETWORK.COM * March 31, 2005* Hello Asterisk-users: I am interested in automatic configurations based on typical legacy phone systems configurations. In other words, after installing Asterisk- I don't want to have to learn the mechanics of the PBX, I just want it to work out of the box. Given that most PBX solutions follow similar configurations, a selection of say 5 or 6 configurations could be selectable in a menu. Obviously some people will need a fully customized solution, and that's fine- I am just looking to reduce my deployment costs. Any ideas? Thanks. *Mark Watson Internet Consultant* Mark Watson Consulting MWCNETWORK.COM 421 Pittsfield Drive Worthington, OH 43085 (614) 923-3929 VOIP Phone (614) 589-2225 Cell [EMAIL PROTECTED] http://www.mwcnetwork.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
cpu load on te4xxp cards is very low, and now that they have echo cancellers as add-ons cards, it will be even lower. I can't speak on hardware compatibility as i never tried a sangoma card. (But i can say that in the last year i've never had an issue with digium cards and we have 8 in use.) The te405p card resolved most incompatibilty issues. /Z signature.asc Description: PGP signature signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Many analog lines
On March 31, 2005 08:53 am, David Hajek wrote: how to use Asterisk where I need to have lets say 40 analog lines. Any ideas? A pair of TE110Ps or a TE405P and an Adit600. This will get you any combination of up to 48 ports, in groups of 8. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN
Hi Maron, Thank you for your answer! I use a simple cisco router 2621XM as call gateway with the following configuration: interface Loopback79 description ALT-VoIP-Gateway ip address 10.xxx 255.255.255.255 h323-gateway voip interface h323-gateway voip id Ldnxxx ipaddr 10.xxx 1719 priority 120 h323-gateway voip h323-id [EMAIL PROTECTED] h323-gateway voip tech-prefix 301 h323-gateway voip bind srcaddr 10.xxx The structure is Sip-phone à SIP à Asterisk as call-manager (extension 399) à H.323 à cisco gatekeeper (extension ) à H.323 à cisco gateway (extension 302) à E1 PSTN Iif I dial now with the Sip-phone: 302 [PSTN number (handy number, .)] I should be able to telephone the the PSTN of the gateway with the extension 302. It works within cisco devices perfectly but not with asterisk. Can you tell me your experiences and practices?? Thanks a lot!! Mario From: Mario Spendier [mailto:[EMAIL PROTECTED] Sent: Donnerstag, 24. März 2005 13:30 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk as Cisco Call-Manager - dial out to PSTN Hi all, Im running Asterisk since two days, and its really one of the phatest software available on the net!!! Respect!!! I have connected Asterisk as a call manager for a cisco gatekeeper. Everything works fine internal, but if I want to ring to a PSTN over another call manager, which is connected over ISDN, I get the following output. Has anyone experience in this or can help me? Im running against closed doors in this problem!!! If I phone over a Cisco call manager it works, so the failure is Asterisk based. -- Executing NoOp(SIP/12345-454d, call for ) in new stack -- Executing Dial(SIP/12345-454d, OH323/ ) in new stack -- H.323 call to with codec alaw -- Called -- H.323 call 'ip$localhost/27230' cleared, reason 24 (Call ended with Q.931 cause) -- Hungup 'OH323/L27230' Thanks a lot!!! Mario ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Realtime - configuration help
Hi, In short : cannot register SIP phone (403 forbidden) In long : I am rather new to asterisk (and linux) One month experience fighting my way in the doc wiki. I worked before with the static '*.conf' files. That worked but I need real-time. I did compile a cvs head 29 mach 2005. MySQL is installed and is running I can access the database from a remote windows pc with access via odbc locally with sql admin sql browser. I created databases following samples explanations in the wiki. extconfig.conf == [settings] sipfriends = odbc,asterisk,sipbuddies voicemail = odbc,asterisk,voicemail_table extensions = odbc,asterisk,extension_table extensions.conf === [from-sip] switch = Realtime/[EMAIL PROTECTED] ; ; voice mail number ; exten = 1999,1,Ringing exten = 1999,2,Wait exten = 1999,3,VoiceMailMain content of sip_buddies table within the asterisk database = (I display only relevant fields) name callerid canreinvite context host mailbox blacky blacky1007 no from-sipdynamic [EMAIL PROTECTED] silver silver1007 no from-sipdynamic [EMAIL PROTECTED] nat typeusernamedisallowallow -1 friend 1007all g729 (I bought licenses) -1 friend 1015all g729 I renamed sip.conf to sip.old Asterisk -vvc shows realtime has started. No sql problems to be seen in log file Realtime mysql status shows : connected to [EMAIL PROTECTED], port 3306 wih username asterisk for xx minutes so ? regards, Shaoul Jacobson VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ACD queue question
Eric would you be so kind as to assist me in setting up an acd que in astersik? ALso I am interested in your domain name rocketgaming.com is that an organization your involved with. On Tue, 29 Mar 2005 21:50:38 -0600, Eric Rees [EMAIL PROTECTED] wrote: I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple authentication
On Mar 31, 2005, at 12:11 AM, Bartosz Wegrzyn - asterisk wrote: Hi, I would like to create extension, so user will have to enter password, and later he will be prompt for a number to call. My config looks like this (ONLY THE PART OF): exten = 888,1,Ringing(), exten = 888,2,wait(2) exten = 888,3,Background,welcome exten = 888,4,Authenticate(1234|a) exten = 888,5,goto(plan,s,1) [plan] exten = s,1,Playback,pls-entr-num-uwish2-call exten = s,2,wait(10) exten = s,3,Dial(SIP/[EMAIL PROTECTED],60,Ttr) exten = s,4,Hangup My understanding was that WAIT() does not listen for input. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ser, asterisk and conferencing
Hi ron, Of course you can make meetme, what you need is a zaptel device or, if you haven't any hardware, the ztdummy device. Install it (google), compile asterisk again, define an extension and it should work, more or less ;-))! Greetings, Mario -Original Message- From: ron [mailto:[EMAIL PROTECTED] Sent: Donnerstag, 31. März 2005 16:07 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ser, asterisk and conferencing Hi List, Can I use asterisk to enable call conferencing? I'm using ser for the UA's to register, can I do something like if they dial a certain digits, it will forward it asterisk and use asterisks meetme feature? can i do meetme using only sip? Sorry for my terms, hope you understand my question. Regards, Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Business Opportunity for Australia
Does anyone want to get involved in sednign traffic from North America to Australia? I want to provide a service were expatriates of Australia that live in NA ? I hope I can post this type of question here? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi looking for missing channel_pvt.h
Hi, I'm trying to compile channel_capi with current Asterisk CVS. Asterisk compiled successfully but channel_capi (patched with all patches needed, as suggested from some nice people on IRC #Asterisk) compilation fails with: app_capiFax.c:34:34: asterisk/channel_pvt.h: No such file or directory I haven't such file on my system! Peraphs patches are for older CVS versions? Thanks Domenico Viggiani ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Many analog lines
You need a T1 card and a channel bank. http://www.voip-info.org/wiki-Asterisk+Channel+Bank Cheers, Jon. On Thursday 31 March 2005 07:53 am, David Hajek wrote: Hi, how to use Asterisk where I need to have lets say 40 analog lines. Any ideas? Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setting SIP to dial PSTN with TDM400P
On Thu, Mar 31, 2005 at 05:05:21PM +0500, Muhammad Haris wrote:
The calls between SIPs and zap phone (fxs) are OK. But 2 issues
cannot be solved:
1. To dial to PSTN via zap phone, the setup in extensions.conf with
the following
exten = _Nxx, 1, zap/1
doesn't work.
I think you need to tell Asterisk what you actually want (Dial) and
tell it the phonenumber, perhaps:
exten = _Nxx, 1, Dial(zap/1/${EXTEN})
Check the wiki for more details about the dial command...
2. When trying using SIP phone to dial PSTN, I got no luck.
Probably same issue...
Hope this helps,
--
Martijn van Oosterhout
Ecomtel Pty Ltd
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Re: [Asterisk-Users] CheckGroup and transfers
On Wed, 30 Mar 2005, C F wrote: I think this bug is what you describe: http://bugs.digium.com/bug_view_page.php?bug_id=0003067 Hope this helps. I think so, but if I'm reading this correctly, the patch is already part of the CVS version? --Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one way audio with X-lite for Linux/Suse 9.2
Hi all, i have installed X-Lite (xlite-linux-22)Suse 9.2 (2.6.8-24) and i have one way audio. The calling number can hear me but i don't hear the called number. Calling my mailbox works fine, i am able to hear my messages. I use a usb handset from Tedas AG. Another strange thing is that the pc, where the X-Lite is installed, start a http connection with brands.xten.net and try to get a file called settings_1103f_9.ini The response is HTTP/1.1 404 Not found. any suggestions... thx in advance __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Asterisk Realtime - configuration help
Hi, (re-posted since I did not see my original one after some time) In short : cannot register SIP phone (403 forbidden) In long : I am rather new to asterisk (and linux) One month experience fighting my way in the doc wiki. I worked before with the static '*.conf' files. That worked but I need real-time. I did compile a cvs head 29 mach 2005. MySQL is installed and is running I can access the database from a remote windows pc with access via odbc locally with sql admin sql browser. I created databases following samples explanations in the wiki. extconfig.conf == [settings] sipfriends = odbc,asterisk,sipbuddies voicemail = odbc,asterisk,voicemail_table extensions = odbc,asterisk,extension_table extensions.conf === [from-sip] switch = Realtime/[EMAIL PROTECTED] ; ; voice mail number ; exten = 1999,1,Ringing exten = 1999,2,Wait exten = 1999,3,VoiceMailMain content of sip_buddies table within the asterisk database = (I display only relevant fields) name callerid canreinvite context host mailbox blacky blacky1007 no from-sipdynamic [EMAIL PROTECTED] silver silver1007 no from-sipdynamic [EMAIL PROTECTED] nat typeusernamedisallowallow -1 friend 1007all g729 (I bought licenses) -1 friend 1015all g729 I renamed sip.conf to sip.old Asterisk -vvc shows realtime has started. No sql problems to be seen in log file Realtime mysql status shows : connected to [EMAIL PROTECTED], port 3306 wih username asterisk for xx minutes so ? regards, Shaoul Jacobson VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Business Opportunity for Australia
I hope I can post this type of question here? No you cannot. Please post this on Asterisk Biz List. That is the right forum. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Walsh Sent: Thursday, March 31, 2005 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Business Opportunity for Australia Does anyone want to get involved in sednign traffic from North America to Australia? I want to provide a service were expatriates of Australia that live in NA ? I hope I can post this type of question here? ___ NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP not working in GUI
[EMAIL PROTECTED] wrote: I have recently installed Asterisk @ 8.0 and loading it fine and setup the ip addressing and change the default password. But when I access the gui from a computer on the network I can pull up the gui but the amp link doesn't work. http://192.168.1.x/admin doesn't work any leads on this. Regards, --- Otis Surratt Jr. / [EMAIL PROTECTED] --- I just loaded and it worked flawlessly for me. Maybe you're going to the wrong IP address? Check your router to see what IP's are registered. JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
hank smith [EMAIL PROTECTED] writes: do you know if it is gtk2? It appears to be: $ ldd xlite-linux-22 ... blah ... libgtk-x11-2.0.so.0 = /usr/lib/libgtk-x11-2.0.so.0 ... blah ... Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma VS. Digium
-Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. If anything puts the dagger to Digium it'll be their own inability to engineer reliable hardware. I appreciate what Digium has done for Asterisk, but reliability expectations for phone equipment are extremely high. I sympathize with people who need hardware that doesn't need to be restarted once a week just to do its job properly. If Digium can't deliver on those reliability expectations, and do it soon, people are going to switch to companies that can. And you know what? I don't blame them. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sms and DDI UK
Is 586671 your ddi number ? I've got a ISDN-32 if that makes any difference Thanks for your help. Julian David Woodhouse wrote: On Thu, 2005-03-31 at 14:10 +0100, Asterisk wrote: Has anyone had any luck with being able to register a DDI with SMS in the UK ? If so, how have you done it ? Anything I send to 0 with a reset message gets sent back to my main ISDN number, even though I have specified a callerid within my DDI range. I did it using 'smsq --queue 586671 0 register' iirc. Certainly I've had both incoming and outgoing calls working on more than just the network number. At the moment they're _all_ on a DDI number, because I've switched to using CAPI and it's using that number for all outgoing calls. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Concurrent Call in Asterisk
Stephen, You should be able to setup what you want. For example, asterisk sip peer will register with your provider. The IP/analog phones will attempt outbound calls which will be sent to this provider. What you need to determine is how your provider bills for the calls. If they bill flat, then you can have 1 user sharing the same account. Otherwise, you may want to check with the provider. HTH Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Sent: Thursday, March 31, 2005 8:56 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Concurrent Call in Asterisk Hi All, Is it possible to have only one SIP account that is shared by several users ? I am currently setting up one asterisk box for a small company (around 7 users). Can all of them make simultaneous call using only one SIP account for termination or I have to setup individual account for all of them (which will be very troublesome on my side as I have to keep reminding them to top up , would be good if I just manage one account) ? Thanks in advance. Stephen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with Music on Hold. Please help
I am having similar issue with Build 1.0.7 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, March 30, 2005 9:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with Music on Hold. Please help Hello everybody, I've run on a problem with music on hold. Asterisk does not play anything. Here is the info: latest Asterisk: Asterisk CVS-HEAD-03/25/05-23:18:57 Asterisk is installed Fedora Core 4 running on AMD 2.0Ghz CPU box with 512 RAM. I took latest zaptel source code, uncommented ztdummy and installed according to instruction from this blog - http://blog.soolid.it/?p=16 I have also compiled and installed Madplay according to same instructions. Zaptel has compiled successfully. Modprobe of zaptel/ztdummy is successful also. However lsmod output shows that USB controller is not used by ztdummy: [EMAIL PROTECTED] src]# lsmod Module Size Used by ztdummy 3924 0 zaptel204676 7 ztdummy ohci_hcd 23765 0 uhci_hcd 31449 0 ehci_hcd 35273 0 Asterisk starts without a problem, the only messages I've receive are following: WARNING[26256]: chan_oss.c:486 soundcard_init: Unable to open /dev/dsp: Device or resource busy == No sound card detected -- console channel will be unavailable ERROR[25489]: cdr_custom.c:135 load_module: Unable to register custom CDR handling Everything else works, but as I said there is no music on hold. sip.conf: In global parameters: musicclass=default ; Sets the default music on hold class for all an extension: [2707] context=default type = friend username = 2707 host = dynamic mailbox = 2707 dtmfmode=rfc2833 nat=no disallow=all allow=ulaw allow=g729 musicclass=default In extensions.conf file: exten = 2707,1,Dial(SIP/2707,35,trHm) ;exten = 2707,2,MusicOnHold() ;exten = 2707,3,MP3Player(/var/lib/asterisk/mohmp3/fpm-sunshine.mp3) exten = 2707,3,voicemail(u2707) exten = 2707,4,Hangup exten = 2707,102,Voicemail(b2707) exten = 2707,103,Hangup musiconhold.conf file: [classes] ;default = quietmp3:/var/lib/asterisk/mohmp3 loud = mp3:/var/lib/asterisk/mohmp3 default = custom:/var/lib/asterisk/mohmp3/,/usr/bin/madplay --mono -R 8000 --output=raw:- But it does not work. when I call extension 2707 on console is following output: Reloading SIP Urgent handler Use EXIT or QUIT to exit the asterisk console -- Executing Dial(SIP/1730-b6a4, SIP/2707|35|trHm) in new stack Urgent handler Urgent handler -- Called 2707 Urgent handler -- Started music on hold, class 'default', on SIP/1730-b6a4 Urgent handler -- SIP/2707-8c7d is ringing Urgent handler -- Stopped music on hold on SIP/1730-b6a4 Any idea what is going wrong? Thanks, NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk-1.0.7 Build - Serious issues
Folks! I want to let everyone know that I have been trying to migrate from 1.0.6 to 1.0.7 last few days and I have come across serious issues in the build 1.0.7. What I found are listed below. I would recommend everyone to hold off any upgrade till the next build. 1)Voicemail - No Audio. Asterisk is not able to stream the voice to the Uas. 0-9 Digit files seem to be missing and Asterisk does not try to say extension numbers for the called user. My guess is all these .gsm files are corrupt and hence you don't hear anything. 2)Music on hold - .MP3 files in the ../mohmp3 and other folders are corrupt. When we tried to play these files using a media player, all we hear is gibberish. 3)DTMF is screwed up. Whatever worked in 1.06 does not work now when we configure this for RFC2833. Has anyone upgraded to 1.0.7 from 1.0.6 and had these issues and been able to find a fix? Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma VS. Digium
My understanding is that to an extent when we buy Sangoma we're putting the dagger to Digium. If anything puts the dagger to Digium it'll be their own inability to engineer reliable hardware. I appreciate what Digium has done for Asterisk, but reliability expectations for phone equipment are extremely high. I sympathize with people who need hardware that doesn't need to be restarted once a week just to do its job properly. If Digium can't deliver on those reliability expectations, and do it soon, people are going to switch to companies that can. And you know what? I don't blame them. I'll second that one for sure. Maybe someone can talk Sangoma into developing a competing TDM04b card? ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
Eric Bishop wrote: True. I think Digium's USA bias is clearly demonstrated by their lack of a BRI ISDN product. Most of the rest of the world use it in abudnace yet Digium do not see fit to service this market because it is not big in the US. very poor... And your EU bias is clearly demonstrated by this. I've never seen a BRI product outside he EU. :-) Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Time sync on PRI
Hi, I am not sure about PRI, but I noticed that * does send date/time info on BRI. Perhaps installing bristuffed Asterisk would solve your problem (bristuff patches libpri, so whatever applies to BRI probably applies to PRI as well). As for syncing PC clock from ISDN line, I haven't noticed any parameter that would control something like that. It would be a nice option, but perhaps it would rise some security issues? Niksa Morten Isaksen wrote: Hi! I have this setup at a customer: PRI - (port 1) TE410P (port 2) - PABC | Asterisk Before the Asterisk part was inserted the customer claims that their PABC automatic changed the clock acourding to daylight saving time from the PRI. Now the customer says that it is not working any more. We are using pri_net signalling up against the PABC and pri_cpe on the other interface. Does Asterisk send time syncronisation on pri_net signalling? Is there a configuration setting that enables that? Is it possible to sync the computer clock with the time from the PRI? The Asterisk server is not connected to a network so NTP is not an option. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Are there online forums instead of this email forum??
Hi, I am new to Asterisk and the first thing I have noticed about Asterisk and Pingtels open PBX's is that they are using this dinosaur method of running forums. It is a real pain getting every message in the forum and essentially keeping my own database of issues. With that said are there any forums that are well used or that might even convert this email in a true forum that is searchable and that doesn't require me downloading every email. Before you go and rant on me go see how Mambo Server does it at http://forum.mamboserver.com. The forums are easy to use and thus are easy to participate in. I use mozilla Thunderbird and I have setup filters and all but it still is a pain to use this outdated email forum. Thanks begin:vcard fn:Chuck Bunn n:Bunn;Chuck org:NetworkDoc LLC adr:;;643 Cougar Loop NE;Albuquerque;NM;87122;USA email;internet:[EMAIL PROTECTED] title:CEO tel;work:505-858-2422 tel;cell:505-264-9221 x-mozilla-html:FALSE url:http://www.networkdoc.com version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] agent and queue autologoff
Guys. While working with agents and queues, you have settings like timeout for agents that dont answer within certain times but I have a question, if you use autologoff, for example, setting the timeout for 15 seconds and autologoff for 30 seconds, then, the agent wont be logged off since you never reach 30 secs acause after 15, the call queue jumps to another agent. Is there a way to make the queue logoff the agent if he doesnt answer 2 calls or maybe make those 30 seconds be a total of unanswered seconds, for example, not answering 2 calls of 15 secs timeout? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel Periodic Reset
Rod Bacon wrote: I have noticed whilst being connected to the console of my * server, that my PRI interface (Digium TE410P) periodically reinitialises itself. This server is currently not actively used, and each time the reset happens the card is idle. Is this normal behaviour, or does it signify a problem? Normal behavior. Helps make sure all the PRI channels are clean. I think ours reset ever hour or so. I also believe this is a configurable parameter in the zaptel.conf -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
Brian Capouch wrote: I'll be glad to stand corrected, but if that assertion is in fact true, we should be careful to do things that actually damage Digium's ability to leverage their development of Asterisk with their hardware sales. It sucks that its such a fine line. On the one had, it is good to have competition. Keeps prices in check, and gets new features out faster. But on the other hand, yes, buying from someone else may say to Digium well, I guess we can stop now that they are buying someone elses cards. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Are there online forums instead of this emailforum??
Hi, I am new to Asterisk and the first thing I have noticed about Asterisk and Pingtels open PBX's is that they are using this dinosaur method of running forums. It is a real pain getting every message in the forum and essentially keeping my own database of issues. With that said are there any forums that are well used or that might even convert this email in a true forum that is searchable and that doesn't require me downloading every email. Before you go and rant on me go see how Mambo Server does it at http://forum.mamboserver.com. The forums are easy to use and thus are easy to participate in. I use mozilla Thunderbird and I have setup filters and all but it still is a pain to use this outdated email forum. It's not a forum, it's a mailing list :-) This might be something to search for you: http://lists.digium.com/pipermail/asterisk-users/2005-March/thread.html You can use google to search the archives. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk::AGI script won't work?
Anyway, you should have this as your first line in the script. #!/usr/bin/perl ___ I had #!/usr/bin/perl5 -w I changed it to #!/usr/bin/perl and now it works. Thanks for the help __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setting SIP to dial PSTN with TDM400P
On Mar 31, 2005 1:05 PM, Muhammad Haris [EMAIL PROTECTED] wrote:
I've setup * with TDM400P w/1 FXS, 1 FXO modules.
I've one analog phone connected to TDM400P FXS module, 1 PSTN line to
one of the FXO module(ZAP) , and IP phone connected to asterisk on
LAN.
The calls between SIPs and zap phone (fxs) are OK. But 2 issues
cannot be solved:
1. To dial to PSTN via zap phone, the setup in extensions.conf with
the following
exten = _Nxx, 1, zap/1
doesn't work.
your line is not correct try this
exten = _NXX,1,Dial(Zap/1/${EXTEN})
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RE: [Asterisk-Users] Are there online forums instead of this emailforum??
I run http://geekgazette.com which has a forum, how-to guides, etc. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Thursday, March 31, 2005 7:27 AM To: Linux - PBX, Asterisk Subject: [Asterisk-Users] Are there online forums instead of this emailforum?? Hi, I am new to Asterisk and the first thing I have noticed about Asterisk and Pingtels open PBX's is that they are using this dinosaur method of running forums. It is a real pain getting every message in the forum and essentially keeping my own database of issues. With that said are there any forums that are well used or that might even convert this email in a true forum that is searchable and that doesn't require me downloading every email. Before you go and rant on me go see how Mambo Server does it at http://forum.mamboserver.com. The forums are easy to use and thus are easy to participate in. I use mozilla Thunderbird and I have setup filters and all but it still is a pain to use this outdated email forum. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk Realtime - configuration help
Shaoul Jacobson - TELLINK wrote: (re-posted since I did not see my original one after some time) You must be more patient than that. Sometimes my posts take a good hour to show. I can access the database from a remote windows pc with access via odbc locally with sql admin sql browser. This tells me that you are using the ODBC driver. Lets keep that in mind. sipfriends = odbc,asterisk,sipbuddies sipfriends is deprecated. You should have seen the warning. This tells me that you did not infact read the wiki. extensions.conf === [from-sip] switch = Realtime/[EMAIL PROTECTED] The [] context cannot be the same as the Realtime context. nat type username disallow allow -1 friend 1007 all g729 (I bought licenses) -1 friend 1015 all g729 Your nat field is wrong. This again..woops..the wiki was never updated for this. OK. Your clear on this one. Your nat field should be varchar(5) using yes, no, never, or route. (wiki updated) Realtime mysql status shows : connected to [EMAIL PROTECTED], port 3306 wih username asterisk for xx minutes Now this is interesting. Above you said you were using ODBC. And all your extconfig stuff says ODBC, but this command here doesn't query via ODBC, it queries MySQL directly. So which is it? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma VS. Digium
Here's an idea, Digium buys Sangoma with the massive amounts of cash they are getting from venture capitalists and just integrate Sangoma designs into their boards. Not sure how Sangoma would feel about this idea though. MATT--- -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Thursday, March 31, 2005 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sangoma VS. Digium Brian Capouch wrote: I'll be glad to stand corrected, but if that assertion is in fact true, we should be careful to do things that actually damage Digium's ability to leverage their development of Asterisk with their hardware sales. It sucks that its such a fine line. On the one had, it is good to have competition. Keeps prices in check, and gets new features out faster. But on the other hand, yes, buying from someone else may say to Digium well, I guess we can stop now that they are buying someone elses cards. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma VS. Digium
On Thu, 31 Mar 2005, Steve Underwood wrote: Eric Bishop wrote: True. I think Digium's USA bias is clearly demonstrated by their lack of a BRI ISDN product. Most of the rest of the world use it in abudnace yet Digium do not see fit to service this market because it is not big in the US. very poor... And your EU bias is clearly demonstrated by this. I've never seen a BRI product outside he EU. :-) Err - hello Steve. From Africa. With BRI. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo on internal SIP
Hi All, On my * server I am getting echo on internal SIP calls. I.E. Sip 2 Sip. Calls going over the T1 via the T100p are fine. I have used ulaw and gsm, gsm has less echo but it is still noticable. All phones are snom 190s. Any ideas on what i can do to cancel this. Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SuperMicro X5DE8-GG Motherboard Goes Kaput after Installing TE410P Card - Yikes!
On Thu, 2005-03-31 at 00:32 -0500, Tim Bass wrote: Hello All, This is my first post. Sorry to post under such sad circumstances. Here is the situation: We installed a TE410P (today) in a SuperMicro 1U server today (Motherboard X5DE8-GG), which was running great until installing this card. After installing the card, the motherboard will not boot (no beeps or indicators) and there is no video output. The fans sign and some of the motherboard lights blink, but like a city with no nightlife, the board is, for all practical purposes, dead.We took the TE410P out and have tried just about very thing under the sun, including clearing the CMOS and, sad to say, the motherboard is still dead with no video out and no beeps. I have had something similar happen with a different card. Does the PSU fan pulse when you attempt to boot? If it is pulsing, you have a short. I have seen standoffs under a motherboard finally touch something that wasn't intended to and the system won't boot. The super micro 1u cases are not very stiff. It is an interesting engineering problem to make something that is hollow stiff with out being able to cross brace. So my suggestion is to pull the board out and maybe hook up to a different PSU with no chance of it shorting out to verify it is ok. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
