Re: [Asterisk-Users] ASTERISK WITHOUT ANY CARD
Andrew Kohlsmith wrote: CAN I USE/COMPILE ASTERISK without any telephone/sound card? I only want to use it as a IP PBX. This is another topic covered quite often. Do we have this in a FAQ/Wiki entry yet? Thank you for the reminder, now it is: http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] files for upgrade cisco 7960 phone
You have to buy a Cisco contract so you can download the files on their site, but here is a link with the explanations, because once you have the good firmware ... there is a way to go : http://www.loligo.com/asterisk/cisco/79xx/ Michael Devenijn DKMA Schaarbeeklei 636 1800 Vilvoorde Tel: +32 2 255 10 19 Fax: +32 2 251 03 12 Van: Carlos Valdes [mailto:[EMAIL PROTECTED] Verzonden: vr 28/11/2003 7:38 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] files for upgrade cisco 7960 phone hi, some on can send me the files for upgrade cisco 7960 phone now is at P0S30202 or where can download ??? thx [EMAIL PROTECTED] winmail.dat
RE: [Asterisk-Users] RFC3389 support incomplete
Just turn off the silence suppresion Van: Jorge Cisneros Flores [mailto:[EMAIL PROTECTED] Verzonden: vr 28/11/2003 5:57 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] RFC3389 support incomplete Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Who i turn off and how i fix this thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
Re: [Asterisk-Users] SIMPLE support in Asterisk?
Leif Madsen wrote: On Thu, 2003-11-27 at 12:03, Mark Spencer wrote: Yea, cause I used both Kphone and Windows messenger, and they successfully registered (and subscribed i think) towards asterisk. Using Kphone I even get a online status on all other users on the asterisk but no interaction with status or IM. So maybe there is some quasi presence avaible? I think it would be a great tool to support IM/Presence. There is so much that can be done with such implementations. SIMPLE could be added within chan_sip, but there is no mechanism within Asterisk to move text from one channel to another *without* the context of a call. *With* the context of a call, we definitely have such a thing (TEXT frames) Some brainstorm notes then: So what's a call for asterisk? * Something that's set up between two endpoints through the dialplan. Simple can send messages within a call, like * A calls B with SIP (INVITE-ACK-ACK) * B sends a URL to A with SIMPLE within the SIP session The problem that we have, if I understand Mark, is that Simple may also be used to send IM without setting up a SIP call (INVITE-ACK-ACK). Like the MWI SIP Notify message from Asterisk. Is that only generated in the relation to a SIP register? Asterisk has some notion of presense (CLI SIP show peers) but not detailed as the normal IM user wants: Presence with some attribute (atoffice, athome, atmistress etc). To get SIMPLE to work within Asterisk, we'll have to: * Add SIMPLE support within the context of a call * Add a new session apart from a call - notification * Add some attributes to presence structure * Add a SUBCRIBE/ACCEPT mechanism - who may subscribe to my whereabouts? * Find programmers that can do this :-) Do the other protocols, MGCP, IAX2, H.323 have any support for text messages? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK WITHOUT ANY CARD
Hcqm wrote: - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Hcqm wrote: CAN I USE/COMPILE ASTERISK without any telephone/sound card? I only want to use it as a IP PBX. Yes, please go ahead. I'm running Asterisk on both LInux and FreeBSD servers without any PSTN or ISDN hardware. I asked because I followup all docs on installation and compile it without problems and edited sip.conf inorder to listen on my ip address on 5060 port. But when I run * the port is not open, no firewalling enabled... My system is a RH9. Any help will be appretiated. Well, enable all debugging and start asterisk with asterisk -vdc Read _ALL_ messages you get, both on the console and in the log files. Most of the time, there's an explanation in there. If not, you should propably tell us more about your specific configuration so we can assist you. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi-line TTS Outbound Dialer
We are working with realspeak and it is a wonderfull product (even in product) it supports up to 20 languages and has aquired a really good prod. stability ! Van: Steve Underwood [mailto:[EMAIL PROTECTED] Verzonden: vr 28/11/2003 4:41 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] Multi-line TTS Outbound Dialer Carl Youngblood wrote: What is EAGI? I will probably use festival for the time being, but I thing that I would eventually like to use ScanSoft's RealSpeak SDK because it is so life-like. Unfortunately our text alerts are fully customizeable, so we can't pre-record them. Beware the likelike TTS, that sucks up thousands of dollars and gets thrown away. RealSpeak is great for demos :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat
[Asterisk-Users] MGCP Support for NAT
Does MGCP transverse NAT? Seeing as the only decent yet cheap IP phone is the Swissvoice, it would be rather helpful. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call waiting disable in sip
Hello, is there a way to disable call waiting in sip? I`m using grandstream 101 and even when the phone is in use I hear ringing in the headset. It is pretty annoying , is there some way to disable this? I cant find anything like it in the grandstream docs. Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line TTS Outbound Dialer
Michael Devenijn wrote: We are working with realspeak and it is a wonderfull product (even in product) it supports up to 20 languages and has aquired a really good prod. stability ! What kind of money we talking for that product? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as SIP Proxy
Hi,
I started evaluating asterisk about a week back. I was trying to configure
asterisk as SIP proxy.
This is the setup that I have now.
I have one linux box running asterisk ( say 192.168.68.15 ) and second box
running partysip (say 192.168.68.6).
I am using SJPhone from windows boxes.
I registered one user to partysip [EMAIL PROTECTED]
and another user to asterisk [EMAIL PROTECTED]
I had this entry in extensions.conf
exten = john,1,Dial(SIP/[EMAIL PROTECTED])
It worked well. I was able to call john.
Now this is what I wanted to achieve.
The other sip server ( here partysip) may have many users registered. It
is not possible to make every user's entry into extensions.conf. Instead,
any mechanism where I can replace 192.168.68.6 with a variable that
represents the 'To' domain will be a great. In simple, I am looking for a
line in extensions.conf that looks like following.
exten = _proxy-.,1,Dial(SIP/${EXTEN:6} @ ${DOMAIN})
*variable DOMAIN is my assumption.
This is the possible soln that I can think off. This, I guess, needs little
hack into chan_sip.c. Is there any other way that simplifies this task?
thanks
-Ranga
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Re: [Asterisk-Users] Asterisk as SIP Proxy
ranga wrote:
I have one linux box running asterisk ( say 192.168.68.15 ) and second box
running partysip (say 192.168.68.6).
Now this is what I wanted to achieve.
The other sip server ( here partysip) may have many users registered. It
is not possible to make every user's entry into extensions.conf. Instead,
any mechanism where I can replace 192.168.68.6 with a variable that
represents the 'To' domain will be a great. In simple, I am looking for a
line in extensions.conf that looks like following.
exten = _proxy-.,1,Dial(SIP/${EXTEN:6} @ ${DOMAIN})
*variable DOMAIN is my assumption.
This is the possible soln that I can think off. This, I guess, needs little
hack into chan_sip.c. Is there any other way that simplifies this task?
Hello!
Never start with considering a hack in the source code, that only creates
a mess for yourself when trying to keep updated. It's easier to read
the documentation :-)
You can create your own variables freely in extensions.conf, see the sample
provided in the Asterisk distribution.
Or:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20readme.variables
http://www.voip-info.org/tiki-index.php?page=Asterisk%20variables
You can also use ENUM for this. See the tips tricks page on the same
server.
The latest CVS version fully supports domain dialling in SIP, so there
should be no problem with calling SIP/${EXTEN:[EMAIL PROTECTED]
If I misunderstood you, please explain a bit more so we can help you.
Regards,
/Olle
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Re: [Asterisk-Users] Resend: Help for oh323
Hi Sathya,
I bet you use OpenH323 v1.12.0.
Go to v1.12.2 and you will be OK.
There isn't anything wrong with your syntax, it's an
OpenH323 issue.
Michael.
SW wrote:
anyone who can shed some light ? Or oh323 is completely dumped and I should
go to chan_h323 ?
-Original Message-
From: SW [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 27, 2003 8:28 AM
To: [EMAIL PROTECTED] Digium. Com
Subject: Help for oh323
Hi Friends,
Hope you would help me out here, I have searched the asterisk
user list for hours and also read the readme and test files that
comes with the driver. I need a very simple scenario. I have SIP
clients and want to use oh323 to dial out to PSTN using a h323 gateway.
a)If I set the extention.conf like this:
exten = _87.,1,Dial(OH323/16.52.153.206)
oh323 dials out (I can ring a netmeeting client at 16.52.153.206).
(b)But if I set it like this, oh323 will not dials out ?
exten = _87.,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED])
In summary what I am trying to achieve is the following;
Lets say Sip user dial 915105418168, then I want 9 to be dropped
and the extension information to be send to the g/w at
16.52.153.206. Isn't exten =
_9x,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED]) is the right
way ?. Why is this not working ?
I must be doing a wrong syntaxt, but couldnt find where I go wrong.
I am attaching the trace for above two cases, please help ?
Cheers
Sathya
Traces for both cases are given below;
0:00.076 OpenH323 Wrapper OpenH323 Wrapper
Version 0.0alpha
0 by inAccess Networks (www.inaccessnetworks.com) on Unix Linux
(2.4.20-8-i686)
at 2003/11/26 20:15:07.622
0:00.078 OpenH323 Wrapper H323Created endpoint.
0:00.078 H323 Cleaner H323Started cleaner thread
0:00.212 OpenH323 Wrapper H323Started listener
Listener[ip$*:1720]
0:00.214H323 Listener:8115c18 H323Awaiting TCP
connections on port 1720
0:00.214 OpenH323 Wrapper H323UDP Binding to
interface: 0.0.0.0:1
0:00.243 OpenH323 Wrapper H323Added capability:
G.711-ALaw-64k{hw} 1
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/hookflash 2
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/basicString 3
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/dtmf 4
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/RFC2833 5
4:00.687 ThreadID=0x4a774440 H323Making call to:
[EMAIL PROTECTED]
4:00.688 ThreadID=0x4a774440 H323Attempt to use
invalid URL [EMAIL PROTECTED]
4:00.688 ThreadID=0x4a774440 H323Could not parse
[EMAIL PROTECTED]
4:00.757 ClearCallT...d:0812ab10 H323Attempt to clear
unknown call
8:14.840 ThreadID=0x4a774440 H323Making call to:
16.52.153.206
8:14.904 ThreadID=0x4a774440 H323Added capability:
G.711-ALaw-64k{hw} 1
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/hookflash 2
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/basicString 3
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/dtmf 4
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/RFC2833 5
8:14.905 ThreadID=0x4a774440 H323Found capability:
G.711-ALaw-64k{hw} 1
8:14.905 ThreadID=0x4a774440 H323Found capability:
UserInput/hookflash 2
8:14.905 ThreadID=0x4a774440 H323Found capability:
UserInput/basicString 3
8:14.906 ThreadID=0x4a774440 H323Found capability:
UserInput/dtmf 4
8:14.906 ThreadID=0x4a774440 H323Found capability:
UserInput/RFC2833 5
8:14.906 ThreadID=0x4a774440 RFC2833 Handler created
8:14.906 ThreadID=0x4a774440 H323Added capability:
G.711-ALaw-64k{hw} 1
8:14.907 ThreadID=0x4a774440 H323Created new
connection: ip$localhost/25259
8:14.908 H225 Caller:80f4688 H225Started call thread
8:15.064 H225 Caller:80f4688 H323TCP Could not connect
to 16.52.153.2
06:1720 (local port=1) - No route to host(113)
8:15.065 H225 Caller:80f4688 H323Clearing
connection ip$localhost
/25259 reason=EndedByConnectFail
8:15.065 H225 Caller:80f4688 H323Call end reason
for ip$localhost
/25259 set to EndedByConnectFail
8:15.066 H225 Caller:80f4688 H225Sending release
complete PDU: ca
llRef=25259
8:15.200 H225 Caller:80f4688 H323Clearing
connection ip$localhost
/25259 reason=EndedByTransportFail
8:15.200 H323 Cleaner H323Cleaning up connections
8:15.201 H323 Cleaner H323Connection
ip$localhost/25259 cl
osing: connectionState=NoConnectionActive
8:15.201 H323 Cleaner H323H323Transport::Close
8:15.201 H323 Cleaner H323
H323Transport::CleanUpOnTerminat
ion for H225 Caller:80f4688
8:15.201 H323 Cleaner H323
RE: [Asterisk-Users] call waiting disable in sip
Anton Yurchenko wrote: Hello, is there a way to disable call waiting in sip? I`m using grandstream 101 and even when the phone is in use I hear ringing in the headset. It is pretty annoying , is there some way to disable this? I cant find anything like it in the grandstream docs. Thanks You need to apply * patch found here: http://bugs.digium.com/bug_view_page.php?bug_id=408 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mailing list archives searchable ?
Hi! I have started development to import the mailinglist archives into a MySQL database and creating a full text search possibility on this. My questions; 1) Is this already done somewhere else ? 2) Is this of interest ? I actually just read someone complaining about this today. I'm sure it'd be a very welcome contribution. The only place that does this is google, and isn't always the best way to search for something specific to the mailing list. For me this is quite sufficient: http://www.mail-archive.com/[EMAIL PROTECTED]/ On the other hand: If someone provided a custom search interface that listed important Asterisk and VoIP keywords and provided a couple of other special features... I am thinking in the direction of a knowledgebase here. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as SIP Proxy
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 28, 2003 2:57 PM
Subject: Re: [Asterisk-Users] Asterisk as SIP Proxy
ranga wrote:
I have one linux box running asterisk ( say 192.168.68.15 ) and second
box
running partysip (say 192.168.68.6).
Now this is what I wanted to achieve.
The other sip server ( here partysip) may have many users
registered. It
is not possible to make every user's entry into extensions.conf.
Instead,
any mechanism where I can replace 192.168.68.6 with a variable that
represents the 'To' domain will be a great. In simple, I am looking for
a
line in extensions.conf that looks like following.
exten = _proxy-.,1,Dial(SIP/${EXTEN:6} @ ${DOMAIN})
*variable DOMAIN is my assumption.
This is the possible soln that I can think off. This, I guess, needs
little
hack into chan_sip.c. Is there any other way that simplifies this task?
Hello!
Never start with considering a hack in the source code, that only creates
a mess for yourself when trying to keep updated. It's easier to read
the documentation :-)
I agree with you. But the issue is, how could I fix the domain name
variable? This should not be static. The target domain changes as per the
choice of the user that is connected through softphone. For example, you are
connected to edvina.net. Now I want to call you from my softphone. I have a
SIP account [EMAIL PROTECTED] This demands me to add your domain in the
configuration of myprovider.com. This server might have a many users and
everybody needs a service extended to the other users connected to other
domains that are running non-asterisk servers. So, everytime a new domain is
requested for dial, the asterisk admin need to add that domain explicitly.
This makes his job tedius.
So, I thought setting DOMAIN variable to the target domain in chan_sip.c
would help. Not sure of complications.
You can create your own variables freely in extensions.conf, see the
sample
provided in the Asterisk distribution.
Or:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20readme.variables
http://www.voip-info.org/tiki-index.php?page=Asterisk%20variables
For some reason, I am seeing SIPDOMAIN blank. Not sure why.
You can also use ENUM for this. See the tips tricks page on the same
server.
The latest CVS version fully supports domain dialling in SIP, so there
should be no problem with calling SIP/${EXTEN:[EMAIL PROTECTED]
I checked it out on 26th of Nov. Any updates in this couple of days towards
this?
If I misunderstood you, please explain a bit more so we can help you.
Its like this: I saw domain dialing in SIP working. When we dial SIP ID from
softphone, asterisk considers the part before '@' as extension. So, we will
need to specifically mention the domain in the call to Dial application.
This is what I wanted to avoid. I would like to pick it from the INVITE
request.
In this case, I can have a standard way of representing the other domain
IDs. For example if I want to call you through my asterisk box, I wil call
you as
sip:[EMAIL PROTECTED]. This way I will not need to mention your domain
name explicitly in the extensions.conf.
Regards,
/Olle
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Re: [Asterisk-Users] MGCP Support for NAT
Hi! Does MGCP transverse NAT? Seeing as the only decent yet cheap IP phone is the Swissvoice, it would be rather helpful. As far as I know: It doesn't since username = IP. However, there is an RFC enhancement proposed to allow MGCP behind NAT, status unkown to me. I did find a note about nat=yes in mgcp.conf on this list but couldn't get it to work (maybe due to my dynamic dial-up IP). As for the Swissvoice: Search the list archive for related msgs - I recently posted my first findings... The e-mail support for IP phones of Swissvoice seems to be asleep, didn't get any reaction from them. Didn't try yet to catch them on the phone, but it appears to be necessary. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as SIP Proxy
Hi!
The other sip server ( here partysip) may have many users registered. It
is not possible to make every user's entry into extensions.conf. Instead,
any mechanism where I can replace 192.168.68.6 with a variable that
represents the 'To' domain will be a great. In simple, I am looking for a
line in extensions.conf that looks like following.
exten = _proxy-.,1,Dial(SIP/${EXTEN:6} @ ${DOMAIN})
*variable DOMAIN is my assumption.
The question is: How does partysip expose its users addresses so that you
can get them into Asterisk? Using AGI with your favourite script language
you could extract those data and set the domain variable.
Cheers, Philipp
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Re: [Asterisk-Users] Re: ASTERISK WITHOUT ANY CARD
CAN I USE/COMPILE ASTERISK without any telephone/sound card? I only want to use it as a IP PBX. YES you can. how about IAX2 trunking? does this work with ztdummy? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MGCP problem
Message: 1 From: Sergi Gabunia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Thu, 27 Nov 2003 12:05:15 +0400 Subject: [Asterisk-Users] MGCP problem Reply-To: [EMAIL PROTECTED] I have VOIP network built with MGCP endpoints.The manufacturer of = endpoints is ASKEY. I downloaded latest Asterisk software and found it = very useful for me. I configured it and it seems taht everything works = OK when I am testing it with one or two endpoints. After that I tried to = move Asterisk to working network and replace existing call manager. It = starts working and calls are proceeding but after a while I could not = hear a dialtone and saw in logs the following: Nov 27 11:40:57 WARNING[10251]: File chan_mgcp.c, Line 2127 = (handle_hd_hf): Unable to create switch thread: Interrupted system call I looked in chan_mgcp.c file and saw that this error occures after = pthread_create functions and it means that this system call was = interrupted permaturely with a signal before it was able to complete.=20 Please, help me to resolve this problem. Best regards, Sergi Gabunia I am using v1.29 of chan_mgcp.c with the askey unit and I can see a similar problem with memory not being released after off hook/on hook transition. Anybody fill me in on what debugging data would be useful in identifying this problem? darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mute button in Grandstream?
Hi John, On Fri, 28 Nov 2003, Anton Yurchenko wrote: Hello, Has anybody been able to get the Mute button work on grandstream? it simply does nothing. Only Hold is avalable, which is not that good. Does the GS even HAVE a mute button? The 101's appear not to. My 100 has MUTE/DEL in the bottom right hand corner. And no, it doesn't work for me either :-) Cheers, Chris. -- _ __ __ _ / __/ / ,__(_)_ | Chris Wilson -- UNIX Firewall Lead Developer | / (_ ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk | \__/_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mute button in Grandstream?
On Fri, 2003-11-28 at 13:24, John Vozza wrote: On Fri, 28 Nov 2003, Anton Yurchenko wrote: Hello, Has anybody been able to get the Mute button work on grandstream? it simply does nothing. Only Hold is avalable, which is not that good. Does the GS even HAVE a mute button? The 101's appear not to. bottom right: MUTE/DEL roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How does Asterisk use CPU?
Hello, I'm trying to figure out what portions of Asterisk need a lot of CPU time. I thought I read somewhere that a Dual P4 2.something will support approximately 80 calls. Is this based on calls that Asterisk is actively doing voice processing for (say, Zap channels and voicemail)? Would a SIP client going through Asterisk and out an IAX channel be CPU intensive if I kept the codec the same throughout the path? I'm probably not asking very clearly, it's awefully late (err, early) but any pointers would be greatly appreciated. Thanks Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Support for NAT
MGCP uses RTP, like SIP and H.323, and can therefore not traverse NAT easily On Fri, 2003-11-28 at 09:18, Andrew Joakimsen wrote: Does MGCP transverse NAT? Seeing as the only decent yet cheap IP phone is the Swissvoice, it would be rather helpful. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk interface with Avaya DS1 Cards?
Hello Everyone.. I just wonder if Asterisk could be interface with the Avaya PBX to its DS1 card partularly the TN2214B? If yes, how should i do it?, just an ordinary T1 cable, or do I need some other equipment? Also, need to know the pin configuration for this. Thanks in advance for the help..:-) joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TEDAS VoIP DECT PABX
Hi, has anyone around here been using/ testing this device? It connects to 1 PSTN and 1 VoIP line, or 2 VoIP lines and allows up to 15 wireless DECT phones to be connected. Pricing? Experiences? Greetings, Philipp http://www.tedas.de/ip_dect.htm Die TEDAS VoIP DECT PABX ist eine drahtlose DECT Telefonanlage für 2 Voice over IP Ports. Das schnurlose Telefon kann abgehende Rufe über einen analogen Anschluss (PSTN) oder über VoIP führen, abhängig von der Konfiguration. Sie kann entweder für den Betrieb mit einer analogen Amtsleitung (PSTN-Port) und einem VoIP-Port oder für zwei VoIP-Ports durch einen mechanischen Schalter konfiguriert werden. Der eingehende Anruf vom Amt oder von VoIP kann von jedem der angeschlossenen DECT Handys entgegengenommen werden. Dieses Produkt vereint ein zwei IP Port- VOIP Gateway mit einer drahtlosen DECT Telefonanlage. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] channel offset between Asterisk and PBX
Hi We interfaced our ASCOTEL PBX to Asterisk. by EuroISDN PRI , DSS1 It works fine on channels 1- 15, but on 17-31 the miststood each other. Asterisk speaks in Timeslots, the PBX in B-channels The signalling is ok, but the bridging is shifted. The first incoming connection is bridged to nirwana also no indication is hearable, calling a second internal subcribes bridges them to the first. The PBX sends a SETUP message with channel identification 30 and Asterisk bridges them to Zap-30, instead of Zap-31. The configuration - Digium TE410p card, set for E1 in zaptel.conf span=1,1,1,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 in zapata.conf signalling = pri_cpe switchtype = euroisdn context = pri1-in pridialplan = unknown channel = 1-15 channel = 17-31 What's wrong? Thanks in advance Roman
Re: [Asterisk-Users] Mute button in Grandstream?
On Fri, 28 Nov 2003, Roy Sigurd Karlsbakk wrote: On Fri, 2003-11-28 at 13:24, John Vozza wrote: On Fri, 28 Nov 2003, Anton Yurchenko wrote: Hello, Has anybody been able to get the Mute button work on grandstream? it simply does nothing. Only Hold is avalable, which is not that good. Does the GS even HAVE a mute button? The 101's appear not to. bottom right: MUTE/DEL ahhh... on my units that button is labeled only DEL and it does not seem to do anything... John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Grandstream BT-100 and
Hey Daniel, Try out the latest CVS and let me know if your problem goes away. We put in a fix for a SIP problem that might be related to this. Thanks, --Greg On Fri, 28 Nov 2003 00:49:20 +0100, Daniel Chabrol wrote: I was successfully using the BT-100 phone with CVS 11/10. Now that I've upgraded to 11/27, I can't place an outbound call. However the phone is registered and works well with inbound calls. Any suggestions will be appreciated. Thank you. Hi! I encounter similar problems. But in my case also incomming calls are not possible. But this might be because of my upgrade to the latest bt-100 beta-firmware (b13p4.22)! Additionally I can't call internal extensions (for example the echo test). Currently i'm using the Asterisk-CVS from 27.11.2003. If i use a softphone (x-lite from x-ten) to connect to * it works perfectly in the same constallation (all with static ip-adresses). Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - Greg Varga Author for RocketryNews http://www.rocketrynews.com CAR # 677 - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line TTS Outbound Dialer
On Fri, 2003-11-28 at 00:42, Carl Youngblood wrote: Or maybe noise would have to last for more than a certain period of time before it triggered another waiting sequence. Like, say, if noise lasts for longer than 2 full seconds or something. That may be fine. Although you may have trouble with some line that also is feeding back echo. That would cause you a bad loop. Like I said, you may wish to limit the number of restarts just in case you end up misdialing a system that is just repeating it's menu. Thanks for the help. All of these telephony issues are fairly new to me. So just for me to understand better, echo is basically something that is difficult to control, right? I mean, if a telco's line has echo, asterisk can't do anything about that, right? While asterisk can do some things about echo, and you have less of a chance of experiencing it if you are on digital lines. Echo can happen that is too much for asterisk to handle. I'm just trying to help you be prepared for what may happen. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Request for debug message in ENUM code
I've been tinkering with ENUM and found that the lack of a debug message in enum.c that says it has actually succeeded in resolving an address is a bit of a nuisance. It makes it difficult to see if failures with ENUM are due to problems with parsing NAPTR records (in enum.c) or mistakes in extensions.conf An extra line of debug information would be much appreciated! Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP-Phones and * audio-files
Hi All, I am a newbie to asterisk, and here is my first problem, where I do not know any further. I have to grandstream BT100 connected to asterisk. Working fine, for calling to each other, and to call via a IAX-Link to the outside. If I try to call the initial demo from the samples.extensions.conf I have nothing to hear. The CLI fine reports: -- Executing Playback(SIP/2209-0260, demo-abouttotry) in new stack -- Playing 'demo-abouttotry' (language 'en') after a few seconds, when I give it up == Spawn extension (demo, 500, 1) exited non-zero on 'SIP/2209-0260' When I call to the voicemail-system with extension 8500, I got also only silence on the phone. What can it bee ?? I tried asterisk with cvs from today (28-11-2003) and with an older version cvs from (19-11-2003) Thanks for any hints something about the hardware: - P4 2.8 GHz - 1 GB RAM - Digium E100P (but not connected at the moment) - Digium TDM400P (but also not connected to devices at the moment) -- Here my additions to the sip.conf disallow=all allow=ulaw allow=alaw allow=g729 allow=g723.1 allow=gsm allow=ilbc allow=speex allow=lpc10 ; my grandstream 102 [2209] type=friend username=2209 secret=nosecretpasswordhere host=dynamic context=demo canreinvite=yes dtmfmode=info qualify=yes disallow=all allow=g723.1 allow=ulaw allow=alaw allow=gsm ; my grandstream 102 [2210] type=friend username=2210 secret=nosecret host=dynamic context=demo canreinvite=yes dtmfmode=info qualify=yes disallow=all allow=ulaw allow=gsm allow=alaw -- in extensions.conf I only added this to lines under section [demo] for testing the calls from gs1 - gs2 exten = 2209,1,Dial(SIP/2209) exten = 2210,1,Dial(SIP/2210) - -- Bye Ernst - Ernst Lehmann Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk interface with Avaya DS1 Cards?
You can connect the Avaya PBX and Asterisk by T1 as well as you need to do configuration in both side Avaya(Trunk group,dial plan...)and *(dial plan...) Howard"Joelson S. Apon" [EMAIL PROTECTED] wrote: Hello Everyone..I just wonder if Asterisk could be interface with the Avaya PBX to its DS1card partularly the TN2214B? If yes, how should i do it?, just an ordinaryT1 cable, or do I need some other equipment? Also, need to know the pinconfiguration for this.Thanks in advance for the help.. :-)joel___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Free Pop-Up Blocker - Get it now
Re: [Asterisk-Users] How does Asterisk use CPU?
On Fri, 2003-11-28 at 06:51, Matthew Asham wrote: Hello, I'm trying to figure out what portions of Asterisk need a lot of CPU time. Mostly codec translations. I thought I read somewhere that a Dual P4 2.something will support approximately 80 calls. Is this based on calls that Asterisk is actively doing voice processing for (say, Zap channels and voicemail)? I think that thread would have been more specific to a task, or just blanket overkill. Would a SIP client going through Asterisk and out an IAX channel be CPU intensive if I kept the codec the same throughout the path? If the codec is the same, then all that is being done is reformating the control protocol. If there is any codec translations, then you would run into some extra overhead. I'm probably not asking very clearly, it's awefully late (err, early) but any pointers would be greatly appreciated. Maybe you would do better to ask more pointed what YOU need help with. If you need help with specing out a machine for your installation, then start specing your installation so we can help. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line TTS Outbound Dialer
Carl Youngblood wrote: What is EAGI? I will probably use festival for the time being, but I thing that I would eventually like to use ScanSoft's RealSpeak SDK because it is so life-like. Unfortunately our text alerts are fully customizeable, so we can't pre-record them. Beware the likelike TTS, that sucks up thousands of dollars and gets thrown away. RealSpeak is great for demos Do you mean that it doesn't work very well in practice, or that it works well but is simply not worth the money? I mean it doesn't work very well. Several of the better TTS engines are now owned by ScanSoft, since they merged with SpeechWorks. I amagine the will rationalise their peoduct lines, and the number will reduce. The very natural sounding TTS packages can be quite hard to understand when you really need to pick out inportant words, like names or addresses. These are just the kinds of things most TTS apps are full of. In practice much less natureal packages, like Eloquence, are more usable. People here the nice natural sounding demos, and get excited. Once you deploy a system for them the complain like mad, and the system gets ripped out. Front end processing is very important for TTS packages - how it will cope with things like dates and currencies in the text stream. Realspeak's is (at least was the last time I tried it) almost useless. Again the demos sound impressive, because they are... well, demos. :-) Try feeding the demo text from each package into all the others. The results can be amusing. Eloquence has the best front end processing, and I believe this has now been plugged into Realspeak (another quite natural sounding product). For me, the joker in the pack is Rhetorical RVoice. The demos sound nice. but I haven't have a chance to really evaluate it. If anyone here has, I'd love to hear their comments. Regards, Steve Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line TTS Outbound Dialer
Brian Capouch wrote: Michael Devenijn wrote: We are working with realspeak and it is a wonderfull product (even in product) it supports up to 20 languages and has aquired a really good prod. stability ! What kind of money we talking for that product? I think it sucks, and it is about $1000 per port. They count ports in a rather maximal way, too. If you have, say, 1000 ports, but a maximum of, say, 10 will be used for TTS at any time they want you to licence 1000 ports of Realspeak. Crazy, but true :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: ASTERISK WITHOUT ANY CARD
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Friday, November 28, 2003 6:50 AM To: Asterisk Users Subject: Re: [Asterisk-Users] Re: ASTERISK WITHOUT ANY CARD CAN I USE/COMPILE ASTERISK without any telephone/sound card? I only want to use it as a IP PBX. YES you can. how about IAX2 trunking? does this work with ztdummy? I was using both IAX2 trunking and MOH before getting my zap devices, and I never had any luck with ztdummy. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323.conf
Hi, this is Cesar Rico I'ma new Asterisk user, I would like toknow how can I develop an application with voice over IP on H323 protocol, I've read all the documentation but I've not found the H323 configuration file,could you send to me a example for this kind offile in order to geta guide on what I have to do. I saw the SIP configuration file, but I don't know if for H323 it is the same, I know that I need the H323 libraries, but the procedure for run the Asterisk is a mystery for me, I have H323 devices (gateway of voice) and I need to register them in an Asterisk server. Let me tell you that I am working on avoice mail application based onmy current H323 devices. Please could you help me sending some documentation on Asterisk over H323? I will appreciate your support so much. Best regards. Cesar Rico CÉSAR AUGUSTO RICO MONDRAGÓN Ingeniero Electrónico ATILA SERVICIOS S.A Cra 11 No 94 - 02 Tel: 57-1-6350785 Movil: 57-1-3102825587 image001.jpg
Re[2]: [Asterisk-Users] How does Asterisk use CPU?
Thanks Steve, that pretty much answers what I wanted to know. I was asking more out of general than for a specific deployment, but if I do have further questions I'll be sure to elaborate. :) Matthew Friday, November 28, 2003, 6:41:23 AM, you wrote: On Fri, 2003-11-28 at 06:51, Matthew Asham wrote: Hello, I'm trying to figure out what portions of Asterisk need a lot of CPU time. Mostly codec translations. I thought I read somewhere that a Dual P4 2.something will support approximately 80 calls. Is this based on calls that Asterisk is actively doing voice processing for (say, Zap channels and voicemail)? I think that thread would have been more specific to a task, or just blanket overkill. Would a SIP client going through Asterisk and out an IAX channel be CPU intensive if I kept the codec the same throughout the path? If the codec is the same, then all that is being done is reformating the control protocol. If there is any codec translations, then you would run into some extra overhead. I'm probably not asking very clearly, it's awefully late (err, early) but any pointers would be greatly appreciated. Maybe you would do better to ask more pointed what YOU need help with. If you need help with specing out a machine for your installation, then start specing your installation so we can help. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting disable in sip
Patrick wrote: On Fri, 2003-11-28 at 09:14, Anton Yurchenko wrote: Hello, is there a way to disable call waiting in sip? I`m using grandstream 101 and even when the phone is in use I hear ringing in the headset. It is pretty annoying , is there some way to disable this? I cant find anything like it in the grandstream docs. Thanks Anton, In sip.conf play with incominglimit= and outgoinglimit=. Brian has fixed whatever wasn't working in cvs: http://bugs.digium.com/bug_view_page.php?bug_id=408 (thanks Brian!). If you want to use this you will need cvs from 11/26 or more recent. what would happend if all operators are busy? would app_queue exit? would it schedule the call to wait and until the number of them reaches the maxlen ( it is defined in queues.conf) ? Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
Hi there! I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. - 3x 6 b-channels. - ~20 Budgetone (and some others) handsets. Can anyone answer these questions: - Will the 3 ISDN cards function correctly in one host? - Will running all 3 cards flat out require particularly beefy hardware? - Will the Grandstream phones provide a good equivilant to professional dedicated PBX phones? (assuming a good network) I have read lots about echo problems and so on, is this an issue? Any help in the matter would be very much appreciated. Thanks in advance! David. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
On Fri, Nov 28, 2003 at 05:02:57PM +, David M. Wilson wrote: - 3x 6 b-channels. - 6 b-channels. Sorry folks! David. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
David M. Wilson wrote: Hi there! I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. As far as I know, AVM only allows a single Fritz!Card PCI in a PC. I /think/ there is a patch out there to allow more than one. Search the archives to find out more. I am sure, you will get better results by putting in an active card. Either AVM or EICON. I have /heard/ the EICON cards are preferable because of the on board echo cancellation -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk behind NAT How to do it.
On Fri, 2003-11-28 at 04:15, Cees de Groot wrote: Leif Madsen [EMAIL PROTECTED] said: outside_addr=216.239.33.100 ; this can also be a FQDN! ie. ; my.domain.com Which might be a problem for dynamic environments, but still nice that someone implemented this patch. I'm sure going to take a look at it - we're currently happy with DIAX calling in to my NAT'ed * setup, but it would be handy to be able to use SIP as well - thanks! U. :) The comment beside that says: this can also be a FQDN! ie. my.domain.com FQDN = Fully Qualified Domain Name This means you can use a hostname like whatever.dynamicip.com However, the variable names have changed since I posted that. They are now: externip; external ip or FQDN localnet; internet ip of asterisk localmask ; subnet mask of internal machine -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iax termination in India
Hi All, Please drop me an email if you can provide Iax termination in India. PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
On Fri, Nov 28, 2003 at 06:15:38PM +0100, Peer Oliver schmidt wrote: As far as I know, AVM only allows a single Fritz!Card PCI in a PC. I /think/ there is a patch out there to allow more than one. Search the archives to find out more. Thanks for the quick response. I'm afraid I was unclear in my original statement - Google pointed me in the direction of this list w.r.t. 3x Fritz cards, and a user who had a problem using that setup. I can't find an answer to his help request. I am sure, you will get better results by putting in an active card. Either AVM or EICON. I have /heard/ the EICON cards are preferable because of the on board echo cancellation I was hoping to reduce costs signifantly by using a fast CPU and cheap passive cards, but if that will result in a noticable loss of quality, then I guess active cards it is then. :) Thanks, David. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
David M. Wilson wrote: Hi there! I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. - 3x 6 b-channels. - ~20 Budgetone (and some others) handsets. Can anyone answer these questions: - Will the 3 ISDN cards function correctly in one host? - Will running all 3 cards flat out require particularly beefy hardware? - Will the Grandstream phones provide a good equivilant to professional dedicated PBX phones? (assuming a good network) I have read lots about echo problems and so on, is this an issue? Any help in the matter would be very much appreciated. Thanks in advance! The driver from AVM only allows one Firtz card in a PC, there is a hack to run two but I don't know about 3.. Save yourself a lot of time and frustration and get a 4 BRI card from Eicon or AVM ( I believe the Eicon is better becasue it has echo cancellation but a little more expensive than the 4BRI AVM card) especially if its for an office install of that many users.. The GS phones are fine and work well, they have a few quirks but these should be fixed on the next firmware upgrade.. If you are running GS phones then the only codec availible to you is the G.711, using this codec you will not need all that powerful a server.. A 1Ghz and above processor with 256+MB of RAM should be fine.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk install / update script - need testers
I have created a script which will install Asterisk from CVS sources with a single command. This was mainly for my own use so that I could do either an install or update without having to enter in all the commands manually. I feel that it is probably stable enough to be released now, but I would like a couple of people to give it a test run, and let me know of any problems with it. Last night, Daniel Quinlan helped me add a couple of options to the file so that now you can stop Asterisk in one of three ways (for instance, if you are updating from CVS, it stops Asterisk before it does the make install, then restarts it after) -c | --conv stop Asterisk with stop when convenient -s | --stop stop Asterisk with stop gracefully -f | --forcestop Asterisk with stop now the default I would also like suggestions from any bash script coders on anything I may not be doing very effectively / wrong (for instance, I don't feel that the way I'm checking to see if the sources already exist is the best way to do it, but it works for me) You can find the code at http://www.hacklocalhost.com/asterisk/_asterisk-update Hopefully someone finds this useful. -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk behind NAT How to do it.
On Fri, 2003-11-28 at 12:23, Leif Madsen wrote: However, the variable names have changed since I posted that. They are now: externip ; external ip or FQDN localnet ; internet ip of asterisk localmask ; subnet mask of internal machine I should also note that they've only changed if you use the latest patch :) http://bugs.digium.com/file_download.php?file_id=448type=bug -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
Hi David, I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. - 3x 6 b-channels. - ~20 Budgetone (and some others) handsets. Can anyone answer these questions: - Will the 3 ISDN cards function correctly in one host? We have done this and it works, using a variant of the hack posted on the website. Points to watch out for: - It's not reliable. We've had Asterisk spontaneously refuse to dial out or accept connections on CAPI until the cards are reset. We don't recommend doing this in production. - You need to be careful with the patch because there are two types of cards, and the patch isn't clever about how it detects them, so either make sure that all your cards are absolutely identical in /proc/pci, or fix the patch. - Will running all 3 cards flat out require particularly beefy hardware? Doesn't seem to. - Will the Grandstream phones provide a good equivilant to professional dedicated PBX phones? (assuming a good network) I have read lots about echo problems and so on, is this an issue? They are cheap and nasty feeling, and not particularly reliable, so I would say no. Cisco 7960 is much better, although more of a pain to get working out of the box, since you need DHCP, TFTP and configuration tools. Cheers, Chris. -- _ __ __ _ / __/ / ,__(_)_ | Chris Wilson -- UNIX Firewall Lead Developer | / (_ ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk | \__/_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QUESTION Ringing Appl.
Hello, I have a problem. When Idial to asterisk with H323 I do not hear ringing applecation (phone rings but i do not hear ringing tone in handset). I have tried with Cisco 2600 H323 and Quintum H323. But when I connect I can hear ringing appl. What can be wrong? Configuration is wrong? Please help! bart
Re: [Asterisk-Users] Asterisk as SIP Proxy
ranga wrote: I agree with you. But the issue is, how could I fix the domain name variable? This should not be static. The target domain changes as per the choice of the user that is connected through softphone. For example, you are connected to edvina.net. Now I want to call you from my softphone. I have a SIP account [EMAIL PROTECTED] This demands me to add your domain in the configuration of myprovider.com. This server might have a many users and everybody needs a service extended to the other users connected to other domains that are running non-asterisk servers. So, everytime a new domain is requested for dial, the asterisk admin need to add that domain explicitly. This makes his job tedius. So, I thought setting DOMAIN variable to the target domain in chan_sip.c would help. Not sure of complications. To call me you don't have to define edvina.net in the asterisk server. Dial(SIP/[EMAIL PROTECTED]) works all right. The problem to fix is when a client, like x-lite, dials sip:[EMAIL PROTECTED]. Asterisk treats this incoming SIP call as a call to oej check the SIPDOMAIN variable to get the edvina.net part and put them back together again. DIAL(SIP/[EMAIL PROTECTED]) should fix it. Just watch out to check if SIPDOMAIN is the realm of your Asterisk server before dialing out. I checked it out on 26th of Nov. Any updates in this couple of days towards this? If I misunderstood you, please explain a bit more so we can help you. Its like this: I saw domain dialing in SIP working. When we dial SIP ID from softphone, asterisk considers the part before '@' as extension. So, we will need to specifically mention the domain in the call to Dial application. This is what I wanted to avoid. I would like to pick it from the INVITE request. That is how it works today. In this case, I can have a standard way of representing the other domain IDs. For example if I want to call you through my asterisk box, I wil call you as sip:[EMAIL PROTECTED]. This way I will not need to mention your domain name explicitly in the extensions.conf. I dial domains from X-lite connected to my ASterisk and it works. If I just enter 10122, X-lite adds the default SIP realm and the server recognizes this as a local extension by checking SIPDOMAIN. If it's not the local SIP realm (like sip:[EMAIL PROTECTED], I add the SIPDOMAIN (as above) and it dials out by checking DNS SRV records. I'll add an example to the Wiki later. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Survey says post your 3.3 volt Mother boards used in PRODUCTION withTE410
I'd like to put up on the wiki the known working 3.3v MotherBoards people are using in production... I am very interested w/ppl with dual te410's with lots of concurrent channels in use Please dont post just your fav spec boards JUST ppl with working stable installs with TE410s, if possible with url, if not the exact board number ill google it when i post to the wiki thxs in advance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323.conf
see /path/to/asterisk/channels/h323/README and then /path/to/asterisk/channels/h323/h323.conf.sample Jeremy McNamara César Rico wrote: Hi, this is Cesar Rico I'm a new Asterisk user, I would like to know how can I develop an application with voice over IP on H323 protocol, I've read all the documentation but I've not found the H323 configuration file, could you send to me a example for this kind of file in order to get a guide on what I have to do. I saw the SIP configuration file, but I don't know if for H323 it is the same, I know that I need the H323 libraries, but the procedure for run the Asterisk is a mystery for me, I have H323 devices (gateway of voice) and I need to register them in an Asterisk server. Let me tell you that I am working on a voice mail application based on my current H323 devices. Please could you help me sending some documentation on Asterisk over H323? I will appreciate your support so much. Best regards. Cesar Rico **CÉSAR AUGUSTO RICO MONDRAGÓN** Ingeniero Electrónico //ATILA SERVICIOS S.A// Cra 11 No 94 - 02 Tel: 57-1-6350785 Movil: 57-1-3102825587 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting disable in sip
- Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 29, 2003 3:34 AM Subject: Re: [Asterisk-Users] call waiting disable in sip what would happend if all operators are busy? would app_queue exit? would it schedule the call to wait and until the number of them reaches the maxlen ( it is defined in queues.conf) ? Hi Anton, Before I submitted the patch to bugtracker to fix this problem, I tested this for both the Dial and Queue apps, and it works as per other channels, ie when all the queue operators are busy, the calling party will stay in the queue until an agent becomes free. All parameters within the queue.conf apply. The only parameter you need to specify in sip.conf is the incominglimit for this to work. For GS phones, set this to 1. By the way, this is no longer a patch as it has been incorporated into the CVS as of 26/11/03. Let me know if you encounter any problems. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Resend: Help for oh323
Michael,
Thanks a bunch, I downloaded from inaccessnetworks.com thinking that it is
the latest :). Ok I will upgrade it. just for the record, following worked.
exten = _87.,1,Dial(OH323/H323:${EXTEN:[EMAIL PROTECTED])
Cheers
Sathya
Date: Fri, 28 Nov 2003 11:28:59 +0200
From: Michael Manousos [EMAIL PROTECTED]
Organization: inAccess Networks
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Resend: Help for oh323
Reply-To: [EMAIL PROTECTED]
Hi Sathya,
I bet you use OpenH323 v1.12.0.
Go to v1.12.2 and you will be OK.
There isn't anything wrong with your syntax, it's an
OpenH323 issue.
Michael.
SW wrote:
anyone who can shed some light ? Or oh323 is completely dumped and I
should
go to chan_h323 ?
-Original Message-
From: SW [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 27, 2003 8:28 AM
To: [EMAIL PROTECTED] Digium. Com
Subject: Help for oh323
Hi Friends,
Hope you would help me out here, I have searched the asterisk
user list for hours and also read the readme and test files that
comes with the driver. I need a very simple scenario. I have SIP
clients and want to use oh323 to dial out to PSTN using a h323 gateway.
a)If I set the extention.conf like this:
exten = _87.,1,Dial(OH323/16.52.153.206)
oh323 dials out (I can ring a netmeeting client at 16.52.153.206).
(b)But if I set it like this, oh323 will not dials out ?
exten = _87.,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED])
In summary what I am trying to achieve is the following;
Lets say Sip user dial 915105418168, then I want 9 to be dropped
and the extension information to be send to the g/w at
16.52.153.206. Isn't exten =
_9x,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED]) is the right
way ?. Why is this not working ?
I must be doing a wrong syntaxt, but couldnt find where I go wrong.
I am attaching the trace for above two cases, please help ?
Cheers
Sathya
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Re: [Asterisk-Users] How does Asterisk use CPU?
if you have 80calls going, its time to think on getting a good dedicated server, switches, for the work and UPS with big batterys also some good power supplie:) Miguel On Fri, 2003-11-28 at 12:51, Matthew Asham wrote: Hello, I'm trying to figure out what portions of Asterisk need a lot of CPU time. I thought I read somewhere that a Dual P4 2.something will support approximately 80 calls. Is this based on calls that Asterisk is actively doing voice processing for (say, Zap channels and voicemail)? Would a SIP client going through Asterisk and out an IAX channel be CPU intensive if I kept the codec the same throughout the path? I'm probably not asking very clearly, it's awefully late (err, early) but any pointers would be greatly appreciated. Thanks Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisck as a Fujistu 9600 VOIP Gateway
Hello, Is there anyone out there who is using asterisk as a VoIP Gateway to a Fujitsu 9600? We have the existing system in place, and I have a mini gateway functioning using a devel kit from digium. I am a systems admin, and know near nothing about the Fujitsu, and could really use some newbie help, if anyone's got such a system running. Many Thanks, Jacob Leaver Senior Systems Engineer ReachONE Internet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk behind NAT How to do it.
Message: 9 From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Organization: http://www.hacklocalhost.com Date: 27 Nov 2003 23:10:42 -0500 Subject: [Asterisk-Users] Asterisk behind NAT How to do it. Reply-To: [EMAIL PROTECTED] Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c version 1.249 (the version the patch was written for) and the latest as of today 1.258. Both work great. Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). Default is 1 - 2 Forward ports 5060 and your RTP range to your internal Asterisk box. For your sip.conf, you need to add three lines: ; sip.conf snippet [general] port=5060 ; make sure you have this line :) inside_net=192.168.1.100; this is the internal ip address of the; asterisk server inside_mask=255.255.255.0 ; internal ip mask. /24 as this example outside_addr=216.239.33.100 ; this can also be a FQDN! ie. ; my.domain.com ; ... plus whatever else you have in your sip.conf Download the patch at: http://bugs.digium.com/file_download.php?file_id=430type=bug Either update your Asterisk or verify you have at least version 1.249 of chan_sip.c: cd /usr/src/asterisk/channels/ cvs status chan_sip.c === File: chan_sip.cStatus: Locally Modified Working revision:1.258 Repository revision: 1.258 /usr/cvsroot/asterisk/channels/chan_sip.c,v While in pwd /usr/src/asterisk/channels/ patch -p0 /path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd Asterisk box, my Free World Dialup number is 18924. Currently online. -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com I can confirm this works for my NAT'd setup as well. Just one comment though that the inside_net variable is your internal subnet address not the asterisk server address. cheers, darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't seem to connect/call fwd network Help!
I have tried everything and still can't place / receive calls from the fwd network. At one point today I was able to call my test machine on the fwd network, I'd answer the call on the test machine (which stated Call Connected), but then the computer I was calling from, through the Asterisk server would give me a 403 Error. I am using sjphone software. I am able to call various extensions with in my network that are setup on the Asterisk server. I can leave and check voice mail with no problem. I just can't seem to connect to anyone outside my network. Below are the error's I received in Asterisk and also my conf files. Any help at all would be GREATLY appreciated! Thanks Dan- Asterisk Prompt error- -- Got SIP response 481 Subscription does not exist back from 192.168.0.105 -- Executing Dial(SIP/78695-eace, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] == No one is available to answer at this time ; SIP Configuration for Asterisk [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to ;externip = 200.201.202.203 ; Address that we're going to put in SIP messages if we're behind a NAT context = sip; Default for incoming calls ;srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes; Turn on support for SIP video ;disallow=all ; Disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc allow=all [fwd.pulver.com] type=friend secret=mypassword username=myfwd# host=fwd.pulver.com [myfwd#] type=friend host=dynamic dtfmode=inband ; Choices are inband,rcf2833, or info context=sip username= myfwd# secret=mypassword mailbox=100 ; Mailbox for message waiting indicator callid=Red myfwd# register =myfwd#:mypassword @fwd.pulver.com/100 [my2ndfwd#] type=friend host=dynamic username=my2ndfwd# secret=mypassword dtmfmode=inband mailbox=101 context=sip callid=Red2 my2ndfwd# register = my2ndfwd#:[EMAIL PROTECTED]/101 Bottom of extensions.conf file [sip] exten = 1,1,Dial(SIP/myfwd#,20,tr) exten = 2,1,Dial(SIP/ my2ndfwd#,20,tr) exten = 100,1,Dial(SIP/ myfwd#,20,tr) exten = 101,1,Dial(SIP/my2ndfwd#,20,tr) exten = 100,2,VoiceMail,u100 exten = 101,2,VoiceMail,u101 exten = 100,102,VoiceMail,b100 exten = 101,102,VoiceMail,b101 exten = 1001,1,Ringing exten = 1001,2,Wait(2) exten = 1001,3,VoicemailMain include = fwd [fwd] exten = _8.,1,Dial,SIP/[EMAIL PROTECTED],tr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXtel down?
Anyone else having timeout problems with IAXtel? Here's the logfile output, user names, passwords, and destination phone numbers have been changed to protect the guilty -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, IAX/someuser:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Calling using options 'exten=1700555;callerid=Steve Rodgers101;language=en;context=iaxtel;username=hwstar;formats=4;capability=14;version=1;adsicpe=2' -- Called someuser:[EMAIL PROTECTED]/[EMAIL PROTECTED] WARNING[1133742896]: File chan_iax.c, Line 1123 (attempt_transmit): Max retries exceeded to host 69.73.19.178 on IAX[69.73.19.178:5036]/11 (type = 6, subclass = 1, ts=1, seqno=0) WARNING[1133742896]: File chan_iax.c, Line 1123 (attempt_transmit): Max retries exceeded to host 69.73.19.178 on IAX[69.73.19.178:5036]/11 (type = 2, subclass = 4, ts=17, seqno=1) -- Hungup 'IAX[69.73.19.178:5036]/11' == No one is available to answer at this time -- Executing Congestion(Zap/1-1, ) in new stack ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXtel down?
On Fri, 28 Nov 2003, Steve Rodgers wrote: Anyone else having timeout problems with IAXtel? Here's the logfile output, user names, passwords, and destination phone numbers have been changed to protect the guilty I just called myself. It worked fine. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXtel down?
Strange, It's back up for me as well... On Friday 28 November 2003 18:49, Joel Maslak wrote: On Fri, 28 Nov 2003, Steve Rodgers wrote: Anyone else having timeout problems with IAXtel? Here's the logfile output, user names, passwords, and destination phone numbers have been changed to protect the guilty I just called myself. It worked fine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Deltathree icomming problem
Hi,
I have a deltathree account and I can place calls
but I can't receive calls. I use Grandstram sip phones. When I call my
deltathree phone # the voicemail is answer :((
I need some help and solutions from the guys who
allready are using deltathree. I search on Internet and I try all types of
configurations... :(
This is my configurations files:
- sip.conf -
[general]port =
5060
; Port to bind tobindaddr =
0.0.0.0
; Address to bind tocontext =
internal
; Default for incoming
callstos=lowdelaydisallow=allallow=ulawallow=gsmallow=alaw
register =
12047440600:[EMAIL PROTECTED]/toti
[iconnect]type=friendport=5060username=12345678secret=1234host=213.137.73.178dtmfmode=inbandcallerid="Chris
Hariga"2407440600
- extensions.conf -
[general]static=yeswriteprotect=yesignorepat =
9
[globals]MYPHONENUMBER=12407440600MYNAME=Chris
HARIGA
[incoming]exten = s,1,Answer()exten
= s,1,Wait(0)exten =
s,2,Dial(SIP/jimSIP/jimofficeSIP/seanSIP/seanhomeSIP/charigaSIP/nadaSIP/laurieSIP/xten|40)exten
= s,3,Voicemail,u100
[internal]ignorepat = 9exten =
toti,1,Dial(SIP/jimSIP/jimofficeSIP/seanSIP/seanhomeSIP/charigaSIP/nadaSIP/laurieSIP/xten|40)exten
= 0,1,Meetme,123exten = _2.,1,SetCallerID(${MYPHONENUMBER})exten
= _2.,2,AbsoluteTimeout(6000)exten =
_2.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],90,r)
Best regards,
Chris HARIGA
