[Asterisk-Users] Re: Re: voip-info: Asterisk record calls
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... LOL I need to read the list completely too before I respond. Hopefully you didn't waste to much time :)) Thank you anyway! -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ominiis Asterisk TAPI driver
I don't think Outlook supports doing a contact lookup from an inbound call. I know Act! Supports that though. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parcina Sent: Tuesday, January 03, 2006 11:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Ominiis Asterisk TAPI driver I have foloved instructions at this web pages http://www.omniis.com/ntsgr/cms/page.asp?688 and now I'm able to call contacts from Outlook. Now I have few questions. When I place a call, my phone rings before * tries to dial out. Is it posible that * first dials out, and when other side picks up, at that moment that my phone rings? Another question, when I recive a phone call, can that contact from Outlook pop-up? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users]MusicOnHold don't start at begin
MusiconHold don't start at begin. What I can doing for setup the musiconhold start at begin?Thanks Fabio Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RPID Issue
Ray Van Dolson wrote: On Sat, Dec 31, 2005 at 10:05:19AM +0100, Olle E Johansson wrote: We're currently planning a new generation of chan_sip that will have a different authentication scheme, not based on the from: header unless it's a local policy to require the From: header to be the same as the Digest auth user name. So to summarize: The Sipura is doing the right thing, but Asterisk can not handle it today, since Asterisk requires a From: user name. You need to disable the caller ID in Asterisk, not in the Sipura. Gotcha. Is there an open bug on this yet? Or should their not be one since it is a planned feature for the future? I'll just continue using my ghetto patch that uses RPID for authentication info as this works in our environment. It's not really a bug, but an effect of the current architecture that is documented and, well, there. Sorry. Will be fixed in a new architecture. Next RPID issue. Our Asterisk server talks to our VoIP provider via a MediaCodes SIP gateway of some sort. They also send us RPID headers. Unfortuantely, in a format that Asterisk does not appear to understand: sip:[EMAIL PROTECTED];party=called;npi=1;ton=2, sip:[EMAIL PROTECTED];party=calling;privacy=off;screen=yes;screen-ind=3;npi=1;ton=2 As you can see it's giving us the called party info first and the calling party info second. get_rpid_num() appears to just check for the first ':' and grab the number immediately afterwards. This is resulting in caller id being set to the called number, which really confuses customers obviously :-) I'm guessing the above is an RFC compliant RPID header and Asterisk's behavior should handle it? I hacked up another patch to address this: http://webdev.digitalpath.net/~rayvd/dist/asterisk/rpid_multiple.patch This works fine as long as we assume that only two entries can be present in the RPID header... Please submit that patch to the issue tracker at bugs.digium.com. Thank you for contributing to Asterisk! /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call-limit kills hints
Senad Jordanovic wrote: Since the device status system relies on it, I rewrote the incominglimit and outgoinglimit into the combined call-limit. The keywords incominglimit and outgoinglimit will be removed, but call-limit will stay. /O Olle/// What happens when it not a simple phone/ATA but a providers trunk which sometimes need different values for IN/OUT channels? Well, groupcount works fine there. Or defining one device for incoming and another for outbound calls to the service provider with different limits. The outgoinglimit never worked, so we haven't had that part working for a long time. It's been disabled in the code since 1.0 I think. /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Regular Crashes
Will that give a fuller bt's than the two below? These were done from
core dumps with asterisk compiled with dont-optimize. I can run asterisk
through gdb but at the moment running with safe_asterisk at least it
automatically restarts after a crash. Though if it will further help
sorting the probem I can run it through gdb.
Regards
Andrew
FIRST TRACE
#0 0x400268b7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
No symbol table info available.
#1 0x0806c146 in ast_mutex_trylock (pmutex=0x672e33fc) at lock.h:597 No
locals.
#2 0x0806175a in ast_queue_hangup (chan=0x672e3330) at channel.c:671
f = {frametype = 4, subclass = 1, datalen = 0, samples = 0,
mallocd = 0, offset = 0, src = 0x0, data = 0x0, delivery = {tv_sec =
0,
tv_usec = 0}, prev = 0x0, next = 0x0}
#3 0x408fc2d9 in __sip_autodestruct (data=0x81be208) at chan_sip.c:1315
p = (struct sip_pvt *) 0x81be208
#4 0x08056c3e in ast_sched_runq (con=0x8172f28) at sched.c:373
current = (struct sched *) 0x8174868
tv = {tv_sec = 1135275568, tv_usec = 989877}
x = 0
res = 1083432672
#5 0x40927e28 in do_monitor (data=0x0) at chan_sip.c:11253
res = 0
sip = (struct sip_pvt *) 0x0
peer = (struct sip_peer *) 0x0
t = 1135275568
fastrestart = 0
lastpeernum = -1
curpeernum = 6
reloading = 0
#6 0x40024b63 in start_thread () from /lib/tls/libpthread.so.0 No
symbol table info available.
#7 0x401ac18a in clone () from /lib/tls/libc.so.6 No symbol table info
available.
SECOND TRACE
#0 0x400268b7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
No symbol table info available.
#1 0x0806c146 in ast_mutex_trylock (pmutex=0x120010c) at lock.h:597 No
locals.
#2 0x0806175a in ast_queue_hangup (chan=0x1200040) at channel.c:671
f = {frametype = 4, subclass = 1, datalen = 0, samples = 0,
mallocd = 0, offset = 0, src = 0x0,
data = 0x0, delivery = {tv_sec = 0, tv_usec = 0}, prev = 0x0, next =
0x0}
#3 0x408fc2d9 in __sip_autodestruct (data=0x81eb518) at chan_sip.c:1315
p = (struct sip_pvt *) 0x81eb518
#4 0x08056c3e in ast_sched_runq (con=0x8172f78) at sched.c:373
current = (struct sched *) 0x8174528
tv = {tv_sec = 1135343875, tv_usec = 693503}
x = 1
res = 0
#5 0x40927e28 in do_monitor (data=0x0) at chan_sip.c:11253
res = 0
sip = (struct sip_pvt *) 0x0
peer = (struct sip_peer *) 0x0
t = 1135343875
fastrestart = 0
lastpeernum = -1
curpeernum = 6
reloading = 0
#6 0x40024b63 in start_thread () from /lib/tls/libpthread.so.0 No
symbol table info available.
#7 0x401ac18a in clone () from /lib/tls/libc.so.6 No symbol table info
available.
Regards
Andrew Gough
Senior Partner
GCD Technologies
Unit 414
Lisburn Enterprise Park
Ballinderry Road
Lisburn
Co Antrim
BT28 2BP
E: [EMAIL PROTECTED]
W: www.gcdtech.com
T: 028 9264 1144
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Robert La Ferla
Sent: 03 January 2006 17:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Regular Crashes
Did you try running * under gdb? When it crashes, do a bt to get a
back trace and post it to the mailing list.
e.g.
% gdb /usr/sbin/asterisk
GNU gdb Red Hat Linux (6.3.0.0-1.84rh)
Copyright 2004 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General Public License, and
you
are
welcome to change it and/or distribute copies of it under certain
conditions.
Type show copying to see the conditions.
There is absolutely no warranty for GDB. Type show warranty for
details.
This GDB was configured as i386-redhat-linux-gnu...Using host
libthread_db library /lib/libthread_db.so.1.
(gdb) run
wait for crash
(gdb) bt
___
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RE: [Asterisk-Users] Is it possible to get caller and callednumberwith Asterisk Manager
I have already looked at Asterisk Events but no one seems to be helpful for my application. I need to get caller and calling number as soon as the communication is started (before call answer) but cdr only log calls after their end. Is there another way to recover these numbers and to associate them to channels? Amaury BOSSÉ De: Giovanni Miano [mailto:[EMAIL PROTECTED] Envoyé: mardi 3 janvier 2006 10:11 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: Re: [Asterisk-Users] Is it possible to get caller and callednumberwith Asterisk Manager No, with Asterisk Manager you can grab Caller and Called ID. See Link, Ring event http://www.voip-info.org/wiki/view/asterisk+manager+events Cheers, Giovanni Miano 2006/1/2, [EMAIL PROTECTED] [EMAIL PROTECTED] : umm - you usually grab it from the cdr...and it works very nicely if you are pushing your cdr into mysql. PaulH - Original Message - From: amaury BOSSE To: asterisk-users@lists.digium.com Sent: Tuesday, January 03, 2006 12:13 AM Subject: [Asterisk-Users] Is it possible to get caller and called numberwith Asterisk Manager Hi list and happy New Year. I working on an application based on Asterisk Manager and I have to recover caller number and called number. Are there manager functions able to do that? If no function already exists, does someone know an issue to resolve my problem? Thanks Amaury ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP security
I have * server that has public IP. Some users with their softphones (and other with hardphones) need to connect to that * server and call out thrue Zap lines. As far as I know when someone tries to authenticate to * server using SIP protocol, he sends data in plain text format. Right? How can I protect * server from snifers? I would like to use SJphone softphone for above purpose. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP security
Tomislav Parcina wrote: I have * server that has public IP. Some users with their softphones (and other with hardphones) need to connect to that * server and call out thrue Zap lines. As far as I know when someone tries to authenticate to * server using SIP protocol, he sends data in plain text format. Right? How can I protect * server from snifers? I would like to use SJphone softphone for above purpose. All SIp authentications are done with MD5 basic digest, not with plain text auth. Plain text auth was remove from SIP a long time ago. /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
I did some research about Asterisk and High Availability and some sort of load balancing. The High Availability issue isnn't much of a problem. I did it with heartbeat en realtime. But the load balancing issue is realy a problem. You want a load balancer to make decisions based on call ID. The call ID is stored in the SIP header (layer 7) and for all I know there are only a few load balancers that can make decisions based on this layer and those load balancers are not SIP aware. So for now I don't think load balancing with *servers could be easily achieved.On 1/4/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Asterisk wrote: In my case I would be using DNS round robin.So a UA would only be registering to one * server at a time.So wouldn't in fact be an active/passive?No. You have said that you want the _other_ servers to be aware of that phone's registration and be able to deliver calls to it directly. Thatwill not work.If you want the other servers to send calls to that phone through theserver it registered with, then yes, that can easily be done. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tijmen van den BrinkWilhelminaweg 46 3441 XC WoerdenTel: 0642233831MSN: [EMAIL PROTECTED]Skype: [EMAIL PROTECTED] SIP:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP security
On Wed, 2006-01-04 at 10:00 +0100, Olle E Johansson wrote: Tomislav Parcina wrote: I have * server that has public IP. Some users with their softphones (and other with hardphones) need to connect to that * server and call out thrue Zap lines. As far as I know when someone tries to authenticate to * server using SIP protocol, he sends data in plain text format. Right? How can I protect * server from snifers? I would like to use SJphone softphone for above purpose. All SIp authentications are done with MD5 basic digest, not with plain text auth. Plain text auth was remove from SIP a long time ago. [long, contains dry math, boring addition follows] to add to this, given the state of MD5 and its 'security' or lack thereof, its a bit over simplistic to just say md5 without adding that its actually 3 md5 hashes... Precomputing is harder (but not impossible) because of the way its done. The nonce makes it a little harder - although the nonce is known even by an attacker ... basically its AUTH1 = md5(username:sip.proxy.com:password) AUTH2 = md5(REGISTER:$URI) final = md5($AUTH1:$NONCE:$AUTH2) AUTH2 is fairly easy to precompute, AUTH1 would change with each attempted password. MD5 has collissions more readily available than advertised. A collission is where two strings result in the same hash ie md5(string1) == md5(string2). this means that it is *possible* (but unlikely under normal circumstances) for someone to guess a 2nd password that would evaluate to the same AUTH1. It would be far more likely that someone would just try to brute force (rainbow tables are not likely to be helpful in this example) the password. MD5 is a fairly fast algorithm, however the fact that it would take effectivly 2 md5 hashes computed you have slowed them down. Zombie machines could be used to speed this up, and it is possible that using MD5 for sip auth it could be brute forced, the probative value of a single account is not likely to justify such an attack. A specific account might, it depends ... Choosing strong passwords helps in this regards (forces the brute force algorithm to be really brute force instead of just a dictionary attack). I would suggest longer passwords, every additional character adds to the overall strength, and basically makes it take longer to exhaust the keyspace. I would also suggest assigned passwords. With sip you set and forget, so users dont need to know what their password really is. If you have 96 characters valid for each position and the password is 8 characters that is 96 to the 8th power, if you goto 14 character passwords (assuming the auth system supports it) you have 96 to the 14th power. A much larger space to attempt to brute force. If you drop the valid chars down to [a-z][A-Z][0-9] omiting all the other common chars on a keyboard at 14 chars you would still be looking at 62^^14 a daunting number. A pc should be able to go through all of those combinations in approx 4,000,000,000,000 years. An exhaustive search of that would be impossible, even though statistically over time on multiple accounts you will do it on average in half that time. Even with a million node zombie network it would still take way longer than the account would be valid for. Now if users get to pick passwords that can be cut dramatically becuase a user sometime is going to use a dictionary based password. With a large enough dictionary and enough zombies an account could in theory be cracked every day. HOWEVER this implies that the attacker could *also* sniff the SIP data to the server. This may be easier than you think, there are a lot of colo facilities and NAPs where people colo systems that dont do good network security simply because it costs money (the routers, switches, etc cant process as much data because they are busy doing the extra compares - NAPs generally are worse about this than colo data centers but YMMV). So the fault in how the attacker is getting the data may not even be yours, but none-the-less they can sniff. If that happens *and* if they have a large enough zombie network *and* they care enough to do this *and* you have weak passwords on your system then there is a potential problem. I think it more likely they would attempt to break into the system that is doing this itself rather than via sniffing SIP auth messages. Side note: doesnt the IAX protocol allow an AUTH request to be sent with no username and will match against all accounts for the supplied password? And if no password is given as well it matches against any user with no password (giving weight to a user named 'guest')? That would seem a far more likely attack than sniffing SIP headers. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed
[Asterisk-Users] Can I call another S0 bus device (BRI) locally without taking 2 channels through Telco provider ?
Hi, I have Asterisk connected to BRI interface in parallel to my ordinary ISDN phone. Can I make internal calls between those two without going through telco provider and taking both voice channels ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call monitoring from 3th phone
is it possible only monitoring call between phone A and B from phone C? -- [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ominiis Asterisk TAPI driver
Tomislav Parcina schrieb: I have foloved instructions at this web pages http://www.omniis.com/ntsgr/cms/page.asp?688 and now I'm able to call contacts from Outlook. Now I have few questions. When I place a call, my phone rings before * tries to dial out. Is it posible that * first dials out, and when other side picks up, at that moment that my phone rings? No. TAPI works this way. It only helps you to get rid of memorizing all kinds of phone number, but you first have to pick up the phone for the dialing to occur. Another question, when I recive a phone call, can that contact from Outlook pop-up? There is at least one third-party addon for Outlook which allows you to just that. Googleing for outlook incoming call popup tapi produces a couple of links. I myself tried ESTOS Procall once and seemed to work okay (mind you, that way before asterisk and is quite a few years ago). HTH -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Raw Hangup messages with IAX2?
Mike McMullen wrote: I found the problem. There was a misconfiguration in the person's firewall that once fixed cleaned everything up. Sorry for the wasted bandwidth. Just for curiosity's sake, what was the misconfiguration? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
Kevin P. Fleming wrote: If the two servers service distinctly separate groups of endpoints, they can share the same table since they won't care about the other server's entries. If the two servers service the same endpoints but in an active/passive arrangement, that would also work. Can the various *RT servers be configured to use different tables so there won't be any conflicts even if there is any client overlap between the servers? What I'm thinking of in this instance is active/active failover. Example: The HA system detects a peer has failed, fences it and then instructs asterisk to take over the registrations in table X that the failed peer was using. How close is this example to reality with *RT? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Dell blade servers
Alistair Cunningham wrote: We've been asked to quote for a large cluster running Asterisk and our ITSP in a box product. The system will be SIP throughout, with mixed codecs. We're considering using Dell blade servers, 1855 or similar, on the grounds that we normally use Dell machines and they work well, but we need higher rack density. Has anyone used these? Any feedback on whether they're good/bad/indifferent? What scalability do you get on simple SIP-SIP forwarding either with or without RTP passing through Asterisk? I would instead recommend the SuperMicro 1U servers - we have had a really great run with these. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] remote sip client fail to register
I attached the logs: any idea? I use SjPhone + STUN and using a dinamyc DSL router with NAT but without any firewall. Sip read: REGISTER sip:172.16.0.4 SIP/2.0 Via: SIP/2.0/UDP 62.0.0.1:49917;rport;branch=z9hG4bK0a03001043bac9d4586a0001 Content-Length: 0 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER From: sip:[EMAIL PROTECTED];tag=1654265612299 Max-Forwards: 70 To: sip:[EMAIL PROTECTED] User-Agent: SJphone/1.60.289a (SJ Labs) 10 headers, 0 lines Urgent handler Sip read: REGISTER sip:172.16.0.4 SIP/2.0 Via: SIP/2.0/UDP 62.0.0.1:49917;rport;branch=z9hG4bK0a03001043bac9d423f50004 Content-Length: 0 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER From: sip:[EMAIL PROTECTED];tag=165427961757 Max-Forwards: 70 To: sip:[EMAIL PROTECTED] User-Agent: SJphone/1.60.289a (SJ Labs) Authorization: Digest username=agallo,realm=asterisk,nonce=153ea438,uri=sip:172.16.0.4,response=9fc986f534bc3ac8ed8901df02cfa94e SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.0.0.1:49917;branch=z9hG4bK0a03001043bac9d423f50004;received=62.0.0.1;rport=5060 From: sip:[EMAIL PROTECTED];tag=165427961757 To: sip:[EMAIL PROTECTED];tag=as580bd78f Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 62.0.0.1:5060 Transmitting (NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 62.0.0.1:49917;branch=z9hG4bK0a03001043bac9d423f50004;received=62.0.0.1;rport=5060 From: sip:[EMAIL PROTECTED];tag=165427961757 To: sip:[EMAIL PROTECTED];tag=as580bd78f Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RBT enable/disable
Hi friends, How i can enable and disable RBT in asterisk for SIP users. We have linksys IP Phones but its give ring to the caller before ringing the called phone. -- Thank You, Code Lover ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.1 Type of Service
Armin Schindler wrote: On Tue, 3 Jan 2006, Steve Beaumont wrote: All, I seem to have a problem with Asterisk 1.2.1. Version 1.0.?? used to allow me to set the Type of Service bits to ef I.e. tos=0xb8 Diffserv EF (Expedited Forwarding) bits. This seems not to be working with 1.2.1:- Jan 3 19:26:26 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:26:36 WARNING[2701] rtp.c: Unable to set TOS to 184 Jan 3 19:33:01 WARNING[18290] rtp.c: Unable to set TOS to 184 Jan 3 19:37:18 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:37:34 WARNING[2701] rtp.c: Unable to set TOS to 184 Jan 3 19:46:25 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:46:38 WARNING[2701] rtp.c: Unable to set TOS to 184 Any help would be appreciated as its playing havoc with the call quality :-( Do you run Asterisk as non-root ? On Linux, only root has the default capability to set the high bits of TOS byte. Armin Armin, Thanks for the response yes, asterisk is running as user 'asterisk':- asterisk 2670 2629 0 08:25 ?00:00:07 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c This is an [EMAIL PROTECTED] distribution Version 2.2 Any idea's how I can runn asterisk as root with [EMAIL PROTECTED] ? Best Regards Steve Beaumont -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.12/220 - Release Date: 03/01/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] USB fxo/fxs devices
Hi, Is there any way to get asterisk to be able to use USB devices like the AUP-03 shown on this website? http://www.chronos.com.tw/products/usb/Skype/skypephone.htm Thanks, Zoltan -- == Geograph (Pty) Ltd P.O. Box 31255 Tokai 7966 South Africa Tel:+27-21-7018492 Fax:+27-86-6115323 Mobile: +27-83-6004028 == ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] confusion about contexts - SER
Hi, Thanks for the reply. What happens is that all users are registered with SER (a sip proxy). I have set SER up so when a user dials 0 followed by a pstn number it will be forwarded to asterisk which will forward the call to a third party pstn gateway. I also use asterisk so that when a user who is registered with ser doesnt answer (sending a 408 cancel response) or is busy (sending a 486 busy response) that the call is forwarded to asterisk voicemail. So therefore at the moment I have a user 2092 which registers with ser and uses the outgoing context in asterisk for pstn access and accesses their voicemail mailbox through the default context. Now I also set it up so that if a user registered with SER dials 20005 it should forwards to asterisk. This should call the context createmenu which creates an IVR menu. What Im confused about is this. I created a user 20005 in sip.conf using context=createmenu. This wasnt working. After reading your post I realized my mistake was that the context that is being called is that of the caller i.e. 2092 as opposed to whom the call is directed at i.e. 20005. Therefore when I changed the context of 2092 to createmenu it worked. BUT how can I set up my sip.conf so that 2092 can use the default, outgoing and createmenu contexts depending on the correct scenario? If someone who is also using SER has any comments, Id also really appreciate it. i.e. [300] type=friend username=300 canreinvite=no context= WHAT GOES HERE?? //createmenu calls the IVR but then outgoing pstn calls dont work, outgoing allows pstn calls but then I cant create a menu etc etc insecure=very ;callerid= voicemail user 1 300 host=dynamic nat=yes dtmfmode=INFO mailbox=300 disallow=all allow=alaw allow=ulaw allow=g723.1 allow=g729 Many thanks, Aisling. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alyed Tzompa Sent: 04 January 2006 00:28 To: asterisk-users@lists.digium.com Subject: re: [Asterisk-Users] confusion about contexts I'm a bit confused on how you get your calls to Asterisk, what I mean is: are you phoning into asterisk via a sip user? in this case, which one?, if not is it iax or though a zap channel? anyway, here some tips: For your first problem it seems it has to do with what I pointed above, check that the user which is dialing into asterisk has the correct context (context=create-menu) with at least type= peer also don't have to retype the allow=codec, disallow=codec, dtmfmode=x for every user, just set it in the general context in sip.conf your second problem think it has to do once again with the firts thing above, and regarding the retyping, I'm afaid I don't know any other way than writing those lines again and again for everyuser. Maybe someone else out thereknows someting else that can help. Don't set many outgoing context for every user in sip.conf! just set one and point all users to that one. If you need your user to have acces to other contextsjust add include = your_context at the end of whatever context you want (btw can add more than oneinlcude's) Alyed --- Hi, Hope someone can help me-Asterisk isnt behaving as I would expect and I think its down to my contexts. There are two things I cant fathom. Firstly I want to record an IVR and so have created a user 20005 and a context called createmenu. I am using SER in front of asterisk so I changed the ser.cfg so that if the user dialled this number it forwards to asterisk. This works fine. The problem is when the invite reaches my asterisk box, asterisk uses the wrong context. It appears to call the outgoing context which is the context used to route calls to my pstn gateway provider. It then trys to execute a Dial command for 20005 which isnt supposed to happen. Secondly SER uses Asterisk for voicemail if a phone doesnt answer after a certain period of time or is busy. This works fine but I have to create an entry for every user in extensions.conf under the [default] context. Can I create a generic entry which would also work to shorten the config file?...Also if I change this and out all the entries under a context voicemail it doesnt work.I have to keep it in defaultThis must obviously be something got to do with Asterisk finding the contexts. I am confused as to how you apply multiple contexts to one user. At the moment nearly each user (besides 20005 and 1234) has a context of outgoing in sip.conf. This is so that they can make outgoing pstn callsBut what if I needed them to use another context in other situations?...Im just confused as to what context should be applied. I have included the relevant parts of my sip.conf and extensions.conf below. I would appreciate any advice as to why these issues are occurring. Many thanks, Aisling. ;sip.conf [general] bindport=5064 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm srvlookup=yes canreinvite=no; autocreatepeer=yes
[Asterisk-Users] Re: Re: connect more the one phone to ONE sip Acoount
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Example of Dial with multiple bindings for one SIP AOR, expanded by FOO:
Dial(${FOO(user1)}) = Dial(SIP/user1/10.1.1.1SIP/user1/10.2.2.2)
That's exactly the solution I've proposed many, many times in the
asterisk-dev mailing list :-)
Will this be implemented before * 1.4 or we have to wait Asterisk 1.4?
--
Tomislav Parcina
[EMAIL PROTECTED]
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[Asterisk-Users] RE: Echo after asterisk has been running for severaldays
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... For this and another issues we reboot our phone servers every week, Saturday 02:00 am. You can do it with croon.weekly. That stopped all the issues This will probably clear the symptoms, but it doesn't solve the problem. Anyway, thank you for your contribution. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.1 Type of Service
On Wed, 4 Jan 2006, Steve Beaumont wrote: Armin Schindler wrote: On Tue, 3 Jan 2006, Steve Beaumont wrote: All, I seem to have a problem with Asterisk 1.2.1. Version 1.0.?? used to allow me to set the Type of Service bits to ef I.e. tos=0xb8 Diffserv EF (Expedited Forwarding) bits. This seems not to be working with 1.2.1:- Jan 3 19:26:26 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:26:36 WARNING[2701] rtp.c: Unable to set TOS to 184 Jan 3 19:33:01 WARNING[18290] rtp.c: Unable to set TOS to 184 Jan 3 19:37:18 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:37:34 WARNING[2701] rtp.c: Unable to set TOS to 184 Jan 3 19:46:25 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:46:38 WARNING[2701] rtp.c: Unable to set TOS to 184 Any help would be appreciated as its playing havoc with the call quality :-( Do you run Asterisk as non-root ? On Linux, only root has the default capability to set the high bits of TOS byte. Armin Armin, Thanks for the response yes, asterisk is running as user 'asterisk':- asterisk 2670 2629 0 08:25 ?00:00:07 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c This is an [EMAIL PROTECTED] distribution Version 2.2 Any idea's how I can runn asterisk as root with [EMAIL PROTECTED] ? Sorry, I don't know anything about [EMAIL PROTECTED] I had the same problem with OpenPBX which is also running as non-root. I managed to inherit the capability 'CAP_NET_ADMIN' which is necessary to set the high TOS bits. For that I patched locally openpbx. I still have no other/better idea. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looping Problem With Call Forwards - Do you have comments on my solution?
On Tue, 2006-01-03 at 12:42 -0600, Brent Torrenga wrote:
I use IP Kall to forward my missed cell phone calls to. This way, if my
phone is off, or out of a service area, calls will go to my * box.
Concurrently, all incoming calls to my * box cause it to dial my local
extensions at home, my extension at work, and my cell phone via NuFone.
Problem: A loop can be created if my cell phone is not on. Say a call comes
into my * box, it uses NuFone to call my cell, my cell forwards the call to
IP Kall, IP Kall to my * box.. You see.
Why not simply something like this:
exten = 123,n,Set(GROUP()=Mobile) ; Set group to the mobile
exten = 123,n,GotoIf($[ ${GROUP_COUNT(${mychan})} 1 ]?next)
exten = 123,n(next),Goto(context,exten,priority) ; Skip over calling
your mobile since there is already another call in progress, ie, send to
VM, or retry your home/office/etc...
exten = 123,n,Dial(IAX2/provider/mobilenum)
On the next line, you could also just play some MoH and wait for 10
seconds, then loop around and try again, etc... but then you are
probably better off using the queue system with agents ...
Hope this helps somewhat...
Regards,
Adam
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[Asterisk-Users] Anybody successfully using vISDN on [EMAIL PROTECTED]
Is there anybody in this group that is using vISDN on an [EMAIL PROTECTED] server? I have a couple of questions, which are quite lengthy, and I do not want to pollute this list of there's no use in asking to begin with! TIA BRgds -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ADM Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Q: How to dial out / transfer calls with manager
On Mon, 2006-01-02 at 09:35 -0800, Don Fanning wrote: From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Action: Originate Channel: SIP/13 -- this should be the first phone you want to ring (your own phone usually) I don't want it to ring a REGISTERED device (SIP/IAX/ZAP) that is on the system. I want it to make a outbound call externally through my VSP and when it's answered, then make another outbound call on another channel. Yes, you obviously need to use a Local channel Channel: Local/contextwithlocalaccess/callernumber That way, the caller (Local channel) gets called first, and after they answer, we then call the destination number via the usual exten parameter. Regards, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Raw Hangup messages with IAX2?
On Wed, January 4, 2006 10:58, Matt Riddell said: Mike McMullen wrote: I found the problem. There was a misconfiguration in the person's firewall that once fixed cleaned everything up. Sorry for the wasted bandwidth. Just for curiosity's sake, what was the misconfiguration? I'd love to know too, as I too see these messages and would like to know how to prevent those... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looping Problem With Call Forwards - Do you have comments on my solution?
On 03/01/06, Brent Torrenga [EMAIL PROTECTED] wrote:
I use IP Kall to forward my missed cell phone calls to. This way, if my
phone is off, or out of a service area, calls will go to my * box.
Concurrently, all incoming calls to my * box cause it to dial my local
extensions at home, my extension at work, and my cell phone via NuFone.
Problem: A loop can be created if my cell phone is not on. Say a call comes
into my * box, it uses NuFone to call my cell, my cell forwards the call to
IP Kall, IP Kall to my * box.. You see.
My solution, and I post this here because I am looking for
comments/improvements to it: When a call comes into my * box, and my * box
dials my cell via NuFone, I will SetGlobalVar ZAPCALLEDTIME to the ${EPOCH}.
Then, whenever a call comes into my * box from IP Kall (aka, any call
forwarded from my cell to my IP Kall), I will take the difference between
the current ${EPOCH} and ${ZAPCALLEDTIME}, compare it to the value 10
(thinking that if it takes less than 10 seconds from the time I forward a
call to my cell, and a forwarded cell call comes into my * box, then it must
be the beginnings of a loop), and if less than 10 to send the call on to be
hungup, or else process it normally.
A simpler solution would be to have the IPKall number forward to a
different extension on your Asterisk server which doesn't include a
call out to the cellphone - and if you use the IPKall number for other
purposes, register yourself another one which you only use for the
cellphone forward.
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
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Re: [Asterisk-Users] Asterisk 1.2.1 Type of Service
On Wed, 2006-01-04 at 12:16 +0100, Armin Schindler wrote: On Wed, 4 Jan 2006, Steve Beaumont wrote: Armin Schindler wrote: On Tue, 3 Jan 2006, Steve Beaumont wrote: All, I seem to have a problem with Asterisk 1.2.1. Version 1.0.?? used to allow me to set the Type of Service bits to ef I.e. tos=0xb8 Diffserv EF (Expedited Forwarding) bits. This seems not to be working with 1.2.1:- Jan 3 19:26:26 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:26:36 WARNING[2701] rtp.c: Unable to set TOS to 184 Jan 3 19:33:01 WARNING[18290] rtp.c: Unable to set TOS to 184 Jan 3 19:37:18 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:37:34 WARNING[2701] rtp.c: Unable to set TOS to 184 Jan 3 19:46:25 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:46:38 WARNING[2701] rtp.c: Unable to set TOS to 184 Any help would be appreciated as its playing havoc with the call quality :-( Do you run Asterisk as non-root ? On Linux, only root has the default capability to set the high bits of TOS byte. Armin Armin, Thanks for the response yes, asterisk is running as user 'asterisk':- asterisk 2670 2629 0 08:25 ?00:00:07 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c This is an [EMAIL PROTECTED] distribution Version 2.2 Any idea's how I can runn asterisk as root with [EMAIL PROTECTED] ? Sorry, I don't know anything about [EMAIL PROTECTED] I had the same problem with OpenPBX which is also running as non-root. I managed to inherit the capability 'CAP_NET_ADMIN' which is necessary to set the high TOS bits. For that I patched locally openpbx. I still have no other/better idea. Armin Assuming your Asterisk boc isn't running any other services why not simply set the TOS with iptables? Rgds Pete ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 wireless and Multicast
On 3 Jan 2006, at 19:10, Francisco Pérez Botella wrote:Hi.I will have to manage From asterisk to clients IP-phones, so biefly the idea is to multiplex voip flows in large packets and multicast them from asterisk/AP to client stations. flows from client stations to asterisk gateway go unicast. I wonder how iax2 protocol will be good for multiplex (trunk) and multicast ??Hmm, it won't be easy.The IAX protocol is not multicast aware, so it is expecting a single ack to each full frame. You will have to do quite a bit of work on the IAX implementationfor it to do the right thing in that area.I'm also not sure I see the advantage of multicast, given that normally phone calls are 1 to 1 connections, (except conferences I suppose).Is it a packet size problem ? http://www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can we dial agents from extensions.conf
On Fri, 2005-12-30 at 20:04 +0530, [EMAIL PROTECTED] wrote:
Thanks a lot Mr. Alexander Lopez for your prompt attension.
I tried the same thing but it wouldnot happen. I use it as:-
exten = 12,1,Dial(Agent/12)
exten = 12,2,Hangup
where agent 12 is configured as :-
agent = 12,12, vivek
After the agent is logged in on extension no12 as follows
Callback Agent '12' logged in on 12
I try to dial 12 from another sip phone and get this:-
-- Executing Dial(SIP/62-c24e, Agent/12) in new stack
-- outgoing agentcall, to agent '12', on 'Local/[EMAIL PROTECTED],1'
-- Called 12
-- Executing Dial(Local/[EMAIL PROTECTED],2, Agent/12) in new stack
Dec 30 14:26:54 NOTICE[13289]: app_dial.c:1011 dial_exec_full: Unable to
create channel of type 'Agent' (cause 17 - User busy)
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack
== Spawn extension (default, 12, 2) exited non-zero on 'Local/[EMAIL
PROTECTED],2'
-- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack
== Spawn extension (default, h, 1) exited non-zero on 'Local/[EMAIL
PROTECTED],2'
== No one is available to answer at this time (1:0/0/0)
-- Executing Hangup(SIP/62-c24e, ) in new stack
== Spawn extension (inoffice, 12, 2) exited non-zero on 'SIP/62-c24e'
-- Executing Hangup(SIP/62-c24e, ) in new stack
== Spawn extension (inoffice, h, 1) exited non-zero on 'SIP/62-c24e'
I am unable to figure out why it is happening like this. They are all in the
same context. Also, the agent is not busy. Also, I wonder why it says Unable
to creat0e chanel of type 'Agent' cause user busy.
Do you have any idea why is it happening so?
I tried to tweak in but was not successful.
You need to use contexts so that the local channel and agent are not in
the same context. eg:
[desks]
exten = 6XX,1,Dial(SIP/${EXTEN}) ; Assumes your sip username is equal
to your extension number
[agents]
exten = 6XX,1,Dial(Local/desks/${EXTEN})
exten = 700,1,AgentCallbackLogin(${CALLERIDNUM},,[EMAIL PROTECTED])
Assumes that your agent id equals your callerid number.
So, somehow, a call is directed to device Agent/601, which will then
call exten [EMAIL PROTECTED] which will then call exten [EMAIL PROTECTED] which
will
then call SIP/601 which is hopefully your SIP phone (ie, device).
Hope that helps
Regards,
Adam
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Re: [Asterisk-Users] Re: Re: connect more the one phone to ONE sip Acoount
Tomislav Parcina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Example of Dial with multiple bindings for one SIP AOR, expanded by FOO:
Dial(${FOO(user1)}) = Dial(SIP/user1/10.1.1.1SIP/user1/10.2.2.2)
That's exactly the solution I've proposed many, many times in the
asterisk-dev mailing list :-)
Will this be implemented before * 1.4 or we have to wait Asterisk 1.4?
The next version is 1.4. No new features is added in 1.2.
I can't promise anything, it depends if someone produces working code or
if someone pays someone to do that :-)
/O
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Re: [Asterisk-Users] Asterisk 1.2.1 Type of Service
On Wed, 4 Jan 2006, Pete Barnwell wrote: On Wed, 2006-01-04 at 12:16 +0100, Armin Schindler wrote: On Wed, 4 Jan 2006, Steve Beaumont wrote: Armin Schindler wrote: On Tue, 3 Jan 2006, Steve Beaumont wrote: All, I seem to have a problem with Asterisk 1.2.1. Version 1.0.?? used to allow me to set the Type of Service bits to ef I.e. tos=0xb8 Diffserv EF (Expedited Forwarding) bits. This seems not to be working with 1.2.1:- Jan 3 19:26:26 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:26:36 WARNING[2701] rtp.c: Unable to set TOS to 184 Jan 3 19:33:01 WARNING[18290] rtp.c: Unable to set TOS to 184 Jan 3 19:37:18 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:37:34 WARNING[2701] rtp.c: Unable to set TOS to 184 Jan 3 19:46:25 VERBOSE[2702] logger.c: == Using TOS bits 0 Jan 3 19:46:38 WARNING[2701] rtp.c: Unable to set TOS to 184 Any help would be appreciated as its playing havoc with the call quality :-( Do you run Asterisk as non-root ? On Linux, only root has the default capability to set the high bits of TOS byte. Armin Armin, Thanks for the response yes, asterisk is running as user 'asterisk':- asterisk 2670 2629 0 08:25 ?00:00:07 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c This is an [EMAIL PROTECTED] distribution Version 2.2 Any idea's how I can runn asterisk as root with [EMAIL PROTECTED] ? Sorry, I don't know anything about [EMAIL PROTECTED] I had the same problem with OpenPBX which is also running as non-root. I managed to inherit the capability 'CAP_NET_ADMIN' which is necessary to set the high TOS bits. For that I patched locally openpbx. I still have no other/better idea. Armin Assuming your Asterisk boc isn't running any other services why not simply set the TOS with iptables? Sure, and even with 'other services' the iptables filter rules should do the job. I just like the idea of having the program doing what ever is necessary without other settings... Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
Tijmen, We use SER for this to load balance across multiple Asterisks. We then use a custom program to monitor the health of the Asterisks and update SER's configuration should one go down. 2 SERs share a single IP address for users to contact using heartbeat. It works well, and we have several customers with it in production. The load balacing isn't perfect, and it can give uneven loads at low capacity, but it gets better as load increases which is where it matters. Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ tijmen van den brink wrote: I did some research about Asterisk and High Availability and some sort of load balancing. The High Availability issue isnn't much of a problem. I did it with heartbeat en realtime. But the load balancing issue is realy a problem. You want a load balancer to make decisions based on call ID. The call ID is stored in the SIP header (layer 7) and for all I know there are only a few load balancers that can make decisions based on this layer and those load balancers are not SIP aware. So for now I don't think load balancing with *servers could be easily achieved. On 1/4/06, *Kevin P. Fleming* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Asterisk wrote: In my case I would be using DNS round robin. So a UA would only be registering to one * server at a time. So wouldn't in fact be an active/passive? No. You have said that you want the _other_ servers to be aware of that phone's registration and be able to deliver calls to it directly. That will not work. If you want the other servers to send calls to that phone through the server it registered with, then yes, that can easily be done. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tijmen van den Brink Wilhelminaweg 46 3441 XC Woerden Tel: 0642233831 MSN: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Skype: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SIP:[EMAIL PROTECTED] mailto:SIP:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-biz] Asterisk on Dell blade servers
Linus, Good point, we'll bear this in mind. Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ Linus Surguy wrote: One thing to be aware of is that Dell blade (as well as many other brand) servers are very heavy beasts. In any deployment with these, check the physical dimensions, check the weight and ensure that it will actually install into the rack that you are using. Also, check the power consumption and heat output and check with your data centre supplier once you know your final rack configuration that it is within their permitted limits. This is essential! Linus Magrathea - Original Message - From: Alistair Cunningham [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Commercial and Business-Oriented Asterisk Discussion asterisk-biz@lists.digium.com Sent: Tuesday, January 03, 2006 5:21 PM Subject: [Asterisk-biz] Asterisk on Dell blade servers We've been asked to quote for a large cluster running Asterisk and our ITSP in a box product. The system will be SIP throughout, with mixed codecs. We're considering using Dell blade servers, 1855 or similar, on the grounds that we normally use Dell machines and they work well, but we need higher rack density. Has anyone used these? Any feedback on whether they're good/bad/indifferent? What scalability do you get on simple SIP-SIP forwarding either with or without RTP passing through Asterisk? -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
On 04/01/06, Alistair Cunningham [EMAIL PROTECTED] wrote: Tijmen, We use SER for this to load balance across multiple Asterisks. We then use a custom program to monitor the health of the Asterisks and update SER's configuration should one go down. 2 SERs share a single IP address for users to contact using heartbeat. I was thinking along the same lines, but for a dynamic setup it should be possible to have SER/OpenSER load balance REGISTER requests according to some strategy/metrics, and then forward INVITEs and other call-related traffic to the 'right' back-end server. Probably lots of reasons why this is too complicated, though Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFax : Change FAX Resolution
Hello all,
Can this be done ?
Would setting the variable FAXRESOLUTION to a appropriate value affect
this change ?
http://www.asteriskguru.com/tutorials/rxfax.html
Variables connected with the application
LOCALSTATIONID - used by to application to identify itself to the remote end
LOCALHEADERINFO - used to generate a header line on each page
REMOTESTATIONID - set by the application, the sender CSID
FAXPAGES - set the number of pages received
FAXBITRATE - set the transmission rate
FAXRESOLUTION - set the resolution
What is the format of setting FAX resolutions ? Currently when i receive
fax's i get the below in the logs.
[Jan 4 18:28:20] DEBUG[8206]:
==
[Jan 4 18:28:20] DEBUG[8206]: Fax successfully received.
[Jan 4 18:28:20] DEBUG[8206]: Remote station id:
[Jan 4 18:28:20] DEBUG[8206]: Local station id:
[Jan 4 18:28:20] DEBUG[8206]: Pages transferred: 1
[Jan 4 18:28:20] DEBUG[8206]: Image resolution: 7700 x 3850
[Jan 4 18:28:20] DEBUG[8206]: Transfer Rate: 9600
[Jan 4 18:28:20] DEBUG[8206]:
==
I tried changing the image resolution by doing below in my dial plan but
still i get the same resolution as above.
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID})
exten = s,2,DBGet(EXTEMAIL=${MACRO_EXTEN}/Email)
exten = s,3,NoOP()
exten = s,4,DBGet(EXTNAME=${MACRO_EXTEN}/Name)
exten = s,5,NoOP()
exten = s,6,DBGet(EXTCOMPANY=${MACRO_EXTEN}/Company)
exten = s,7,SetVar(FAXRESOLUTION=6000 x 3000)
exten = s,8,rxfax(${FAXFILE}.tif|debug)
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(7)
exten = s,105,SetVar(EXTNAME=Unknown)
exten = s,106,Goto(7)
exten = s,107,SetVar(EXTCOMPANY=Unknown)
exten = s,108,Goto(7)
Any ideas ?
TIA
dushyanth
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Re: [Asterisk-Users] Asterisk on Dell blade servers
- Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 04, 2006 10:11 AM Subject: Re: [Asterisk-Users] Asterisk on Dell blade servers Alistair Cunningham wrote: We've been asked to quote for a large cluster running Asterisk and our ITSP in a box product. The system will be SIP throughout, with mixed codecs. We're considering using Dell blade servers, 1855 or similar, on the grounds that we normally use Dell machines and they work well, but we need higher rack density. Has anyone used these? Any feedback on whether they're good/bad/indifferent? What scalability do you get on simple SIP-SIP forwarding either with or without RTP passing through Asterisk? I would instead recommend the SuperMicro 1U servers - we have had a really great run with these. -- Cheers, Matt Riddell If you're considering 1U rack servers then also look at the Gigabyte SR147L (P4 Socket 478) or SR157L (P4 LGA 775) we've deployed over 150 and had only one PSU failure in 3 years (do use the Western Digital RAID edition SATA drives though). Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 wireless and Multicast
El Miércoles, 4 de Enero de 2006 12:28, tim panton escribió: On 3 Jan 2006, at 19:10, Francisco Pérez Botella wrote: Hi. I will have to manage From asterisk to clients IP-phones, so biefly the idea is to multiplex voip flows in large packets and multicast them from asterisk/AP to client stations. flows from client stations to asterisk gateway go unicast. I wonder how iax2 protocol will be good for multiplex (trunk) and multicast ?? Hmm, it won't be easy. The IAX protocol is not multicast aware, so it is expecting a single ack to each full frame. You will have to do quite a bit of work on the IAX implementation for it to do the right thing in that area. I see, maybe I could redirect at network layer unicast--multicast addresses/group and give back a false single ack at that point. On the other side (client side). I need some like a virtual trunk where each station recieves the full frame and stealth the payload it needs for the user/phone(s) it serves. I could at client station redirect traffic from multicast to unicast interface address and serve the full frame to iax2 at client station, silently dropping the acks they give back. I'm also not sure I see the advantage of multicast, given that normally phone calls are 1 to 1 connections, (except conferences I suppose). That maybe true for wired but wireless in infraestructure mode there's a point of distribution (the AP) that even can police and pool in a pseudo TDM, I mean all the traffic in the subnet is going to pass trought that point Is it a packet size problem ? It's a capacity problem first, it's an avoidance of collisions too. wireless is a shared medium (radio) and minimizing overhead without latency penalty will be important. I think that in a radio system broadcast is for free capacity and overhead is not. http://www.westhawk.co.uk/ -- Francisco J. Pérez Botella ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems Upgrading to 1.2.1 on Fedora 3
On Tue, Jan 03, 2006 at 06:43:16PM -0500, Michael Stearne wrote: and when I try to update from binary: [EMAIL PROTECTED] ~]# rpm -Uvh asterisk-1.2.1-15.rhfc3.at.i386.rpm warning: asterisk-1.2.1-15.rhfc3.at.i386.rpm: V3 DSA signature: NOKEY, key ID 66534c2b error: Failed dependencies: libpri.so.1 is needed by asterisk-1.2.1-15.rhfc3.at.i386 libspandsp.so.0 is needed by asterisk-1.2.1-15.rhfc3.at.i386 libtonezone.so.1 is needed by asterisk-1.2.1-15.rhfc3.at.i386 obviously you need to install the packages that include them (libpri, zaptel/libtonezoe, spandsp). And you need a version for Asterisk 1.2 . Do you use apt/yum for installing those? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compilation of OpenH323 libraries under CYGWIN...
Hi everybody, was anybody able to compile whole OpenH232 package under CYGWIN? I was not able to link plugins... Regards everybody and thank you Mauro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax : Change FAX Resolution
Dushyanth Harinath wrote:
Hello all,
Can this be done ?
Would setting the variable FAXRESOLUTION to a appropriate value affect
this change ?
http://www.asteriskguru.com/tutorials/rxfax.html
Variables connected with the application
LOCALSTATIONID - used by to application to identify itself to the remote end
LOCALHEADERINFO - used to generate a header line on each page
REMOTESTATIONID - set by the application, the sender CSID
FAXPAGES - set the number of pages received
FAXBITRATE - set the transmission rate
FAXRESOLUTION - set the resolution
What is the format of setting FAX resolutions ? Currently when i receive
fax's i get the below in the logs.
[Jan 4 18:28:20] DEBUG[8206]:
==
[Jan 4 18:28:20] DEBUG[8206]: Fax successfully received.
[Jan 4 18:28:20] DEBUG[8206]: Remote station id:
[Jan 4 18:28:20] DEBUG[8206]: Local station id:
[Jan 4 18:28:20] DEBUG[8206]: Pages transferred: 1
[Jan 4 18:28:20] DEBUG[8206]: Image resolution: 7700 x 3850
[Jan 4 18:28:20] DEBUG[8206]: Transfer Rate: 9600
[Jan 4 18:28:20] DEBUG[8206]:
==
I tried changing the image resolution by doing below in my dial plan but
still i get the same resolution as above.
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID})
exten = s,2,DBGet(EXTEMAIL=${MACRO_EXTEN}/Email)
exten = s,3,NoOP()
exten = s,4,DBGet(EXTNAME=${MACRO_EXTEN}/Name)
exten = s,5,NoOP()
exten = s,6,DBGet(EXTCOMPANY=${MACRO_EXTEN}/Company)
exten = s,7,SetVar(FAXRESOLUTION=6000 x 3000)
exten = s,8,rxfax(${FAXFILE}.tif|debug)
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(7)
exten = s,105,SetVar(EXTNAME=Unknown)
exten = s,106,Goto(7)
exten = s,107,SetVar(EXTCOMPANY=Unknown)
exten = s,108,Goto(7)
Any ideas ?
TIA
dushyanth
FAX resolutions are set by the sender, not the receiver. The receiver
can reject modes and resolutions it does not support - super fine, grey
scale, colour etc - but it can't force the sender to use a higher
resolution.
Steve
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[Asterisk-Users] Re: OT: XML Content Manager for Cisco 79XX Phones
In article Pine.LNX.4.44.0601031441190.9209-10 @dulles1.contactgga.com, [EMAIL PROTECTED] says... For anyone interested, our company released a PHP/MySQL based content manager for the Cisco 79XX series IP Phones compatible with the SIP load yesterday. It's available via: http://www.sourceforge.net/projects/open79xxdir I like it. Thank you! -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can i compile Asterik on Fedora 4 x86 64 and which hardware could you support ?
Hi everyone , I want to compile asterisk on Fedora 4 64 bit edition . is there any one has experience on compiling asterisk on 64 bit linux? .. could you suggest me cpu , main board for x86 64 architecture? Followings are my sample configuration : Dual-Core AMD BOX OPTERON CPU 280-2400MHz S2895UA2NRF S2895_SCSI DUAL AMD OPTERON DDR,PCI-X, GBE, PCI-E ,AC97 AUDIO, DIGITAL SPDIF,FIREWIRE,SCSI mainboard 4 GB ddr ram Fedora 4 for 64 bit i am afraid of hardware compatibility problem and i have no experience on x86 64 architecture.. so i need help On asterisk there will be 250 simultaneous calls( for out band calls only) .. Just using SIP protocol and g279 and g 723 codecs... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: call monitoring from 3th phone
Asterisk cmd ZapBarge ZapBarge(channel) Lets you listens to the conversation on a specified Zap channel, or prompts if one is not specified. You can hear them, but they can't hear you. No indication is given to the other parties that their call is being listened to. http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OT: XML Content Manager for Cisco 79XX Phones
In article Pine.LNX.4.44.0601031441190.9209-10 @dulles1.contactgga.com, [EMAIL PROTECTED] says... For anyone interested, our company released a PHP/MySQL based content manager for the Cisco 79XX series IP Phones compatible with the SIP load yesterday. It's available via: http://www.sourceforge.net/projects/open79xxdir Permisions, permisions... You could use some good documentation. When I try to create database this is what I get: Could not retrieve information from database, please go back and make sure database information is correct and that Apache has writing permissions. Help please. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Raw Hangup messages with IAX2?
Subject: Re: [Asterisk-Users] Raw Hangup messages with IAX2? Mike McMullen wrote: I found the problem. There was a misconfiguration in the person's firewall that once fixed cleaned everything up. Sorry for the wasted bandwidth. Just for curiosity's sake, what was the misconfiguration? -- Cheers, Matt Riddell Hi Matt, The person at home had their IAX2 ports forwarded to the wrong IP address. (Though they swore they didn't!) ;-) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk on Dell blade servers
I would instead recommend the SuperMicro 1U servers - we have had a really great run with these. Which 1U models have you found work best? Do you know if ABE has been tested or certified on any SuperMicro platforms? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] suddenly iax calls don't work anymore
Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial(Zap/2-1, IAX2/username:password@sip.coco-connect.de/number) in new stack -- Called username:password@sip.coco-connect.de/number -- Call accepted by 62.180.50.221 (format g729) -- Format for call is g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1758 ast_set_write_format: Unable to find a path from gsm to g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? ... Gerald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Raw Hangup messages with IAX2?
On Wed, January 4, 2006 14:53, Mike McMullen said: Subject: Re: [Asterisk-Users] Raw Hangup messages with IAX2? Mike McMullen wrote: I found the problem. There was a misconfiguration in the person's firewall that once fixed cleaned everything up. Sorry for the wasted bandwidth. Just for curiosity's sake, what was the misconfiguration? -- Cheers, Matt Riddell Hi Matt, The person at home had their IAX2 ports forwarded to the wrong IP address. (Though they swore they didn't!) ;-) Mike Hmzzz... That's not my problem though, so I quess I'll need to investigate further! :-( Thanks for the info tho! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Entry level IP phone
Hi, Happy New Year to all of you! I was wondering what would be the best recommended entry level IP phone that works well with * if buying say around 10 handsets. Linksys spa-941 and the grandstreamgxp-2000look like good phones but I'm open to recommendations Cheers, Reggie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Using *RT for HA purposes was: [Asterisk-Users] RealtimeMultipleAsterisk boxes, iaxusers
I think I have 4 options. 1, Modify chan_sip.c to update a new field in sipusers realtime table with the status of the sip peer/user. Then use agi to dial sip calls. Check the status field if OK then dial the fullcontact from the sip table. If not goto voicemail or where ever else I want the call to go.. The UA would only register to one server, so only one server *should* be writing to the database. (If not the code will be modified to do so.) Other servers would be read only from my AGI. 2, Use Asterisk management interface to find the status of the sip peer. Then dial fullcontact if peer is active. Should be easy to implement. Problem is I would have to actively poll each server in the farm. 3, Use SER as the sip router and asterisk as an application/media server. Then all sip UA would register to the SER. Should scale higher, but does add a level of complexity. 4, Continue to use IAX trunk to dial the other switch. Then hope that realtime has been improved by the time I need the 3rd server. It is a failover but not a load balancer. Any thoughts? Am I completely off here? Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Wednesday, January 04, 2006 7:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Using *RT for HA purposes was: [Asterisk-Users] RealtimeMultipleAsterisk boxes, iaxusers On 04/01/06, Alistair Cunningham [EMAIL PROTECTED] wrote: Tijmen, We use SER for this to load balance across multiple Asterisks. We then use a custom program to monitor the health of the Asterisks and update SER's configuration should one go down. 2 SERs share a single IP address for users to contact using heartbeat. I was thinking along the same lines, but for a dynamic setup it should be possible to have SER/OpenSER load balance REGISTER requests according to some strategy/metrics, and then forward INVITEs and other call-related traffic to the 'right' back-end server. Probably lots of reasons why this is too complicated, though Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: suddenly iax calls don't work anymore
In article [EMAIL PROTECTED], Gerald Dachs [EMAIL PROTECTED] wrote: Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial(Zap/2-1, IAX2/username:password@sip.coco-connect.de/number) in new stack -- Called username:password@sip.coco-connect.de/number -- Call accepted by 62.180.50.221 (format g729) -- Format for call is g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1758 ast_set_write_format: Unable to find a path from gsm to g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? ... Looks like your end is offering g729 as a supported encoding format, and the other end is choosing it, but your system doesn't actually have the G.729 codec. In iax.conf, you can either add disallow=g729, or probably even better, put disallow=all and then specific allow lines for the codecs you want to support, such as alaw, gsm and so on. Don't forget a reload in order to act upon the changes. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] suddenly iax calls don't work anymore
Gerald Dachs wrote: Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial(Zap/2-1, IAX2/username:password@sip.coco-connect.de/number) in new stack -- Called username:password@sip.coco-connect.de/number -- Call accepted by 62.180.50.221 (format g729) -- Format for call is g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1758 ast_set_write_format: Unable to find a path from gsm to g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? ... Gerald Someone will probably correct me, but it looks like you are trying to use the g729 codec for your calls (or coco-connect.de is forcing you to use g729), but this requires a license from Digium and is not installed on your machine. Try using a different codec if possible or, if you do have a g729 license try re-installing the codec and re-activating it. I think this may solve the problem. But as I say, someone may correct me - I may be completely wrong about this. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unknown digits
Is this normal to have entries like this on a PRI? Jan 3 10 43 22 DEBUG[7341] Exception on 83, channel 69 Jan 3 10 43 22 DEBUG[7341] Got event Event 131126(131126) on channel 69 (index 0) Jan 3 10 43 22 DEBUG[7341] DTMF Down '6' Jan 3 10 43 22 DEBUG[7341] Exception on 83, channel 69 Jan 3 10 43 22 DEBUG[7341] Got event Event 262198(262198) on channel 69 (index 0) Jan 3 10 43 22 DEBUG[7341] Pulse dial '6' Jan 3 10 43 22 DEBUG[7341] Exception on 83, channel 69 Jan 3 10 43 22 DEBUG[7341] Got event Event 131121(131121) on channel 69 (index 0) Jan 3 10 43 22 DEBUG[7341] DTMF Down '1' Jan 3 10 43 23 DEBUG[7341] Exception on 83, channel 69 Jan 3 10 43 23 DEBUG[7341] Got event Event 262193(262193) on channel 69 (index 0) Jan 3 10 43 23 DEBUG[7341] Pulse dial '1' There is a DTMF and Pulse referrence for each. Usually 1 or 2 digits per call. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FYI new aricle on asteisk
Got my latest Linux magazine (www.linux-magazine.com) and fetured is asterisk in home network. I've also been in contact with Novel/SUSE about their asterisk pakages. Reinhard Max the maintainer. He has hinted at new packages for SUSE 10. The current ones work well (in production) however he is unsure about the new zaptel intergration but I'm keeping my fingers crossed! -- A.G. (Tony) NicholsI.S. Manager ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime Multiple Asterisk boxes, iaxusers
Mike Fedyk wrote: Can the various *RT servers be configured to use different tables so there won't be any conflicts even if there is any client overlap between the servers? Yes, but I'm not sure how you'd manage failover in that situation then. What I'm thinking of in this instance is active/active failover. Example: The HA system detects a peer has failed, fences it and then instructs asterisk to take over the registrations in table X that the failed peer was using. There is not currently any way to accomplish that, unless you do it in the database itself. If your database supports updatable views, then each server could actually be connected to a view that provided only the desired rows out of the master table, and the failover process could rebuild the view for the new 'active' server. It'd be a bit ugly, but not horrible. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk on Dell blade servers
LJ wrote: Which 1U models have you found work best? Do you know if ABE has been tested or certified on any SuperMicro platforms? It has not. If you wish to see that happen, contact SuperMicro and arrange for them to supply some systems for certification testing; we'd be happy to see that happen. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] suddenly iax calls don't work anymore
Thanks, that helped Gerald On Wed, 04 Jan 2006 14:39:51 + Faris Raouf [EMAIL PROTECTED] wrote: Gerald Dachs wrote: Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial(Zap/2-1, IAX2/username:password@sip.coco-connect.de/number) in new stack -- Called username:password@sip.coco-connect.de/number -- Call accepted by 62.180.50.221 (format g729) -- Format for call is g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1758 ast_set_write_format: Unable to find a path from gsm to g729 Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 NOTICE[23409]: channel.c:1791 ast_set_read_format: Unable to find a path from g729 to slin Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? Jan 4 10:06:42 WARNING[23409]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from IAX2 (20)? ... Gerald Someone will probably correct me, but it looks like you are trying to use the g729 codec for your calls (or coco-connect.de is forcing you to use g729), but this requires a license from Digium and is not installed on your machine. Try using a different codec if possible or, if you do have a g729 license try re-installing the codec and re-activating it. I think this may solve the problem. But as I say, someone may correct me - I may be completely wrong about this. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
Peter Bowyer wrote: I was thinking along the same lines, but for a dynamic setup it should be possible to have SER/OpenSER load balance REGISTER requests according to some strategy/metrics, and then forward INVITEs and other call-related traffic to the 'right' back-end server. Probably lots of reasons why this is too complicated, though One being that it must be the device that NAT phones register with that delivers calls to them. Otherwise, the NAT device sees a packet coming from an unknown IP address and drops it (for common types of NAT such as restricted cone). Since SER needs to deliver calls, it really needs to be SER that accepts REGISTERs and holds the registration information. The Asterisks then send calls from phones to the SER heartbeat address for delivery. This is what we do in our ITSP in a box product. It gives us full redundancy and failover with the registration capacity of SER and the features of Asterisk. For very large systems, it's possible to have SER redirect (with load balancing) REGISTERs to a set of SERs so that NAT devices know about the machines their phones are registered on, but this takes great care to get right in all cases. Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Start recording after call started
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In Asterisk v1.2.1 check the featuremap section of the features.conf file. You also need to add the w or W option to your Dial cmd where appropriate. So with the feature mapping below pressing *1 would start recording. [featuremap] blindxfer = #1; Blind transfer, default is # disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer I need to dail *1 to quickly. Can that be changed? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
On 04/01/06, Alistair Cunningham [EMAIL PROTECTED] wrote: Peter Bowyer wrote: I was thinking along the same lines, but for a dynamic setup it should be possible to have SER/OpenSER load balance REGISTER requests according to some strategy/metrics, and then forward INVITEs and other call-related traffic to the 'right' back-end server. Probably lots of reasons why this is too complicated, though One being that it must be the device that NAT phones register with that delivers calls to them. Otherwise, the NAT device sees a packet coming from an unknown IP address and drops it (for common types of NAT such as restricted cone). Yes, that's the sort of reason I was thinking of :-) I guess you could NAT the whole cluster behind a single IP with some fancy firewall/router rules Since SER needs to deliver calls, it really needs to be SER that accepts REGISTERs and holds the registration information. The Asterisks then send calls from phones to the SER heartbeat address for delivery. And if a lot of the calls are SIP-SIP, I guess - why bother Asterisk with them at all... This is what we do in our ITSP in a box product. It gives us full redundancy and failover with the registration capacity of SER and the features of Asterisk. Sounds good. For very large systems, it's possible to have SER redirect (with load balancing) REGISTERs to a set of SERs so that NAT devices know about the machines their phones are registered on, but this takes great care to get right in all cases. Yeah - I knew this was harder than it looked :-) Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Ominiis Asterisk TAPI driver
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I don't think Outlook supports doing a contact lookup from an inbound call. I know Act! Supports that though. To bad that tipicly user doesn't change his faworite mail reader because of increased functionality in VoIP ;)) Thank you anyway. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAx/g729 client for MAC
There is an idefisk for mac available for alpha testers, contact me off list for a copy. Zoa tim panton wrote: On 27 Nov 2005, at 13:43, Chris Mason (Lists) wrote: Is there a good quality stable (not free) IAX2 client for MAC? I have a client wants to travel and make calls and I want to avoid the SIP blocking that is a problem for travellers. I have heard good things about http://www.loudhush.ro/ But haven't used it (yet). I couldn't get any of the other IAX2 clients to be stable on the MAC. I very much doubt you will find a g729 client for the mac. The thing with g729 is that you have to license quite large numbers of clients just to get the patent holders to talk to you. I'd go for GSM, it is nearly as effecient as g729, people are used to the way it sounds (from mobiles) and it is patent free. My traveling (windows carrying) users are getting on fine with IAX2 over GSM. Tim. http://www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how do i play a prerecorded message in the middle of a conversation ?
as the subject says, suppose i want to do the phone-equivalent of cutpaste on a messagging program, i.e. play back a prerecorded file in the middle of a conversation, is there anything that lets me do the trick by working on the dialplan, or i should go and write my own res_features trick ? Ideally i would like to come up with something like dialng *5xxx where xxx is the message number i want to playback. cheers luigi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 wireless and Multicast
On 4 Jan 2006, at 13:28, Francisco Pérez Botella wrote:El Miércoles, 4 de Enero de 2006 12:28, tim panton escribió: On 3 Jan 2006, at 19:10, Francisco Pérez Botella wrote: Hi.I will have to manage From asterisk to clients IP-phones, so bieflythe ideais to multiplex voip flows in large packets and multicast them fromasterisk/AP to client stations. flows from client stations to asteriskgateway go unicast. I wonder how iax2 protocol will be good formultiplex(trunk) and multicast ?? Hmm, it won't be easy.The IAX protocol is not multicast aware, so it is expecting a singleack to eachfull frame. You will have to do quite a bit of work on the IAXimplementationfor it to do the right thing in that area. I see, maybe I could redirect at network layer unicast--multicast addresses/group and give back a "false" single ack at that point.On the other side (client side). I need some like a "virtual trunk" where each station recieves the full frame and "stealth" the payload it needs for the user/phone(s) it serves. I could at client station redirect traffic from multicast to unicast interface address and serve the full frame to iax2 at client station, silently dropping the acks they give back.yes, but you need to ensure that only one client station sends an ack, orthat the server station can cope with multiple acks. I'm also not sure I see the advantage of multicast, given that normallyphone calls are 1 to 1 connections, (except conferences I suppose). That maybe true for wired but wireless in infraestructure mode there's a point of distribution (the AP) that even can police and pool in a pseudo TDM, I mean all the traffic in the subnet is going to pass trought that pointSure, but that isn't any different from any asterisk server connected to an ethernet (except in speed) (Wasn't the pre-cursor of ethernet a radio based net in Hawaii ?). You aren't saving very much capacity,as IAX miniframes have a low overhead. You would probably do better torun IAX over the lowest level protocol you can get at (i.e. lose IP and UDPheaders and go straight to the packet radio level). Is it a packet size problem ? It's a capacity problem first, it's an avoidance of collisions too.wireless is a shared medium (radio) and minimizing overhead without latencypenalty will be important. I think that in a radio system broadcast is for free capacity and overhead is not. Yep, but apart from the headers you won't be saving any actual payload bytes, unless more than client is listening to the same stream at the same time.As for collisions, I see a (nasty) problem that trunking might cause:Many IAX clients use the incoming audio stream as a timing source to the outgoing one. In a Trunked/multicast situation you'd have all yourclients replying in sync - which would cause collisions, since they would all reply at once. You would have to impose some delay on theclient side to ensure they didn't. Easier not to trunk I'd say.T. http://www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Ominiis Asterisk TAPI driver
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... No. TAPI works this way. It only helps you to get rid of memorizing all kinds of phone number, but you first have to pick up the phone for the dialing to occur. Well I have to get use to press speaker bottun :)) There is at least one third-party addon for Outlook which allows you to just that. Googleing for outlook incoming call popup tapi produces a couple of links. I myself tried ESTOS Procall once and seemed to work okay (mind you, that way before asterisk and is quite a few years ago). I'll try some of them. Can anybody else sugest something? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AMP: Losing backslash characters in config files
I've just started using AMP and found that I have a problem with escaped characters in config files. In particular, I have a custom config item that needs a semicolon in... SetVar(_ALERT_INFO=info=auto-answer;delay=1) To get the part of the line after the ; to be accepted by Asterisk as a non-comment it needs to be escaped with a backslash, but I have found that I need to put two backslashes in to get this through AMP. Subsequent to this, when AMP displays the page it has stripped one of the backslashes. The outcome of this is that if you view and then update a working config file it breaks. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Start recording after call started
On Wed, January 4, 2006 15:45, Tomislav Parcina said: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In Asterisk v1.2.1 check the featuremap section of the features.conf file. You also need to add the w or W option to your Dial cmd where appropriate. So with the feature mapping below pressing *1 would start recording. [featuremap] blindxfer = #1; Blind transfer, default is # disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer I need to dail *1 to quickly. Can that be changed? Try experimenting with this: [general] featuredigittimeout = 1000 ; Max time (ms) between digits for ; feature activation. Default is 500 HTH! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FC3 or FC1 (or something else?)
From what ive read on this list and the wiki, centos 4.x has issues with the TE110P card ( a lot of people having issues after first reboot).Would 3.5 be better (I know [EMAIL PROTECTED] uses this) Am I right in saying that OS's with the 1.6 kernel still require a lot more tinkering than those with the 1.4 kernel ?? Does anybody know what Digiums stance on OS is , I remember speaking to them about 6 months ago and they were recommending a 1.4 kernel version of Debian. Are there any specific disadvantages to running 1.4 kernel ??, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: 04 January 2006 02:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FC3 or FC1 (or something else?) Michael Stearne wrote: I am having trouble with FC3. After doing a yum update (of 1264 packages) I still cannont compile 1.2.1 from source: make[1]: `libedit.a' is up to date. make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline' make[1]: Entering directory `/usr/src/asterisk-1.2.1/db1-ast' make[1]: `libdb1.a' is up to date. make[1]: Leaving directory `/usr/src/asterisk-1.2.1/db1-ast' make[1]: Entering directory `/usr/src/asterisk-1.2.1/stdtime' make[1]: *** No rule to make target `/usr/lib/gcc/i386-redhat-linux/3.4.2/include/stddef.h', needed by `localtime.o'. Stop. make[1]: Leaving directory `/usr/src/asterisk-1.2.1/stdtime' make: *** [stdtime/libtime.a] Error 2 and when I try to update from binary: [EMAIL PROTECTED] ~]# rpm -Uvh asterisk-1.2.1-15.rhfc3.at.i386.rpm warning: asterisk-1.2.1-15.rhfc3.at.i386.rpm: V3 DSA signature: NOKEY, key ID 66534c2b error: Failed dependencies: libpri.so.1 is needed by asterisk-1.2.1-15.rhfc3.at.i386 libspandsp.so.0 is needed by asterisk-1.2.1-15.rhfc3.at.i386 libtonezone.so.1 is needed by asterisk-1.2.1-15.rhfc3.at.i386 I have compiled from source 1.0.9 without problem on this machine. Any ideas why my attempts are now failing? To use the binary, it appears that you need libpri. Look at http://www.asterisk.org (top right hand side) Did you try getting the latest source via svn subversion? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAx/g729 client for MAC
On 27 Nov 2005, at 13:43, Chris Mason (Lists) wrote: Is there a good quality stable (not free) IAX2 client for MAC? I have a client wants to travel and make calls and I want to avoid the SIP blocking that is a problem for travellers. I have heard good things about http://www.loudhush.ro/ But haven't used it (yet). I couldn't get any of the other IAX2 clients to be stable on the MAC. I've been using Loudhush and really like it. jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Start recording after call started
Tomislav Parcina wrote: I need to dail *1 to quickly. Can that be changed? Speed dial button or programmable button for your IP phone works... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SUSE 10.1
I have been told the next version of SUSE will contain the 1.2.1 build. I am unsure if the zaptel module will be ready -- but I have hight hopes! Per my last post... 10.0 is working very well in production -- including the auto updates.-- A.G. (Tony) NicholsI.S. Manager ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom Firmware 5.0.
Old Ringer 2 4 will be available as 9 10 (in addition to the existing melodies) in Version 5.1 to be released in a few days. Its better than wasting bandwidth downloading such a custom melody, as Ringer2 seems so popular. Hope that will suffice... Regards, Usman. Message: 13 Date: Tue, 3 Jan 2006 10:05:35 -0600 From: Joe Pukepail [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] snom Firmware 5.0. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I agree, I liked the old ringtone 2 also (just a beep), I use it at my desk, If I'm there I can pick it up and it wasn't obnoxious enough to disturb others. Please email it to me if you get it in the format needed. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Raw Hangup messages with IAX2?
Hmzzz... That's not my problem though, so I quess I'll need to investigate further! :-( Thanks for the info tho! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ When I googled on asterisk raw hangup I came across a few postings regarding the problem. They seemed to all point to the wrong info/ip address being presented by the client's router/ firewall. This is what made me go back and check again on the person's router who swore all was set correctly. HTH, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Ominiis Asterisk TAPI driver
CounterPath's X-Pro Tapi softphone has this I think? http://www.xten.com/index.php?menu=X-Series (select the EU region) I think they have a trial...downloading it now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parcina Sent: Wednesday, January 04, 2006 8:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Ominiis Asterisk TAPI driver In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... No. TAPI works this way. It only helps you to get rid of memorizing all kinds of phone number, but you first have to pick up the phone for the dialing to occur. Well I have to get use to press speaker bottun :)) There is at least one third-party addon for Outlook which allows you to just that. Googleing for outlook incoming call popup tapi produces a couple of links. I myself tried ESTOS Procall once and seemed to work okay (mind you, that way before asterisk and is quite a few years ago). I'll try some of them. Can anybody else sugest something? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceMailMain Pass Mailbox
I have a extension 981 setup for entering VoiceMailMain: exten = 981,1,VoiceMailMain,([EMAIL PROTECTED]) exten = 981,2,HangUp() I want to pass the calling extension to the context (extension and mailbox numbers are the same). This dosen't seem to work. I get this in the console: Asterisk Ready.*CLI -- Executing VoiceMailMain(SIP/2504-ba66, [EMAIL PROTECTED]) in new stack -- Playing 'vm-login' (language 'en') Any ideas? Thanks!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMailMain Pass Mailbox
use ${CALLERIDNUM} instead of [mailbox]
I have a extension 981 setup for entering VoiceMailMain:
exten = 981,1,VoiceMailMain,([EMAIL PROTECTED])
exten = 981,2,HangUp()
I want to pass the calling extension to the context (extension and mailbox
numbers are the same).
This dosen't seem to work. I get this in the console:
Asterisk Ready.
*CLI -- Executing VoiceMailMain(SIP/2504-ba66, [EMAIL PROTECTED]) in
new stack
-- Playing 'vm-login' (language 'en')
Any ideas?
Thanks!!
___
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___
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Re: [Asterisk-Users] Echo cancellation
Asterisk performs echo cancellation for all incoming and outgoing calls through the T1/PRI card. However, there are some things that canstill cause echo's. We had a similar situation a with a setup much like yours. We switched to the MG2 echo canceller which helped quite a bit. But finally, after testing everything else we possibly could, swapping out PRI cards, changing every concevable setting related to echo, and having the phone company out about a dozen times to check the PRI, we ended up switching companies from Telepacific to Sprint and the echo problem, according to the users, completely went away. However, a couple things you may want to look at is the audio gain settings in your zapata.conf and make sure they're not set too high, maybe even try dropping them down a bit. And try the MG2 echo canceller if you haven't already, it seems to provide the best results on PRI's, at least as far as our testing has gone. Erick On 1/3/06, Aaron Daniel [EMAIL PROTECTED] wrote: We currently have about 60 cisco 7940's, which were converted from ciscocall manager to be used for asterisk.We're running 1.2.1 stable on 4systems (primary server, backup server, gateway, and voicemail).Thephone lines come into the gateway on a digium te405p.The problem we'rehaving is that the 7940's are echoing on outgoing calls, and I'm not sure what else to try (I did just recompile zaptel with a different echocanceler to see if it would help), but I seem to remember asteriskdoesn't do echo cancellation on outgoing calls.I researched a bit on qos and latency, but there's maybe a 10ms latency between the phone andthe outgoing line, so I ruled that out pretty quick.Any help isgreatly appreciated.AaronOn Tue, 2006-01-03 at 21:26 -0800, Erick Baum wrote: Can you provide some details about the system, what version of Asterisk, what kind of phones, what kind of phone lines, etc. Erick On 1/3/06, Aaron Daniel [EMAIL PROTECTED] wrote: I've got a slight problem with echo.Basically, most of the outgoing phone calls on our system echo, but as far as I can tell, the incoming echo has been relatively fixed, with just a bit of work left to do on it.I read somewhere that asterisk doesn't echo cancel on outgoing calls, am I wrong in that assumption, and if I am, what else can be done when the echo training and echo cancel tapping isn't working? Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMailMain Pass Mailbox
[mailbox] does not exist
use
exten = 981,1,VoiceMailMain,(${CALLERID(num)}@usvm)
this
is provided that your callerid settings in your sip, iax, and zap configs are
correct and relect the extension calling.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest
BeckSent: Wednesday, January 04, 2006 11:43 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users]
VoiceMailMain Pass Mailbox
I have a extension 981 setup for entering VoiceMailMain:
exten = 981,1,VoiceMailMain,([EMAIL PROTECTED])
exten = 981,2,HangUp()
I want to pass the calling extension to the context (extension and
mailbox numbers are the same).
This dosen't seem to work. I get this in the console:
Asterisk Ready.*CLI -- Executing
VoiceMailMain("SIP/2504-ba66", "[EMAIL PROTECTED]") in new
stack -- Playing 'vm-login' (language
'en')
Any ideas?
Thanks!!
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[Asterisk-Users] chan_oh323.so freeze my box on unload
Hi im running several gentoo servers with Asterisk, only using IAX2 and SIP. Recently we decided to implement h323. All the necessary dependences for oh323-0.7.3 were installed by portage (package manager of Gentoo distro), including openh323, pwlib etc. The module is successfully loaded (load chan_oh323.so) but when asterisk is stopped (stop now) or the oh323 module is unloaded (unload chan_oh323.so) the computer just freezes, the keyboard stop responding, cannot open more ssh sesssions, but the computer still responds 'pings', but no service seems to be responding (web, ssh) -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] M0n0Wall traffic shaping rules
Hi all, Anyone got any VoIP traffic shaping rules for m0n0wall that they could let me look at please? Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_oh323.so freeze my box on unload
My apologies. My fingers just pressed enter before i completed my email to you. Just a few more indications. As i said, the box respond pings, but anything else does not work, no ssh, no web, no dns, no email. NMAP told me that all the services ports are open (22, 25, 52, 80) even the H323 1720 port. All of this is kind of hard to debug since no logs are saved. Any ideas? On 1/4/06, Moises Silva [EMAIL PROTECTED] wrote: Hi im running several gentoo servers with Asterisk, only using IAX2 and SIP. Recently we decided to implement h323. All the necessary dependences for oh323-0.7.3 were installed by portage (package manager of Gentoo distro), including openh323, pwlib etc. The module is successfully loaded (load chan_oh323.so) but when asterisk is stopped (stop now) or the oh323 module is unloaded (unload chan_oh323.so) the computer just freezes, the keyboard stop responding, cannot open more ssh sesssions, but the computer still responds 'pings', but no service seems to be responding (web, ssh) -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] remote sip client fail to register
From: sip:[EMAIL PROTECTED] ;tag=165427961757Max-Forwards: 70To: sip:[EMAIL PROTECTED]Keep Attention: y our softphone is sending internal ipCheers,Giovanni Miano2006/1/4, Antonio Gallo [EMAIL PROTECTED] :I attached the logs: any idea?I use SjPhone + STUN and using a dinamyc DSL router with NAT but without any firewall.Sip read:REGISTER sip:172.16.0.4 SIP/2.0Via: SIP/2.0/UDP62.0.0.1:49917;rport;branch=z9hG4bK0a03001043bac9d4586a0001 Content-Length: 0Contact: sip:[EMAIL PROTECTED]:5060Call-ID: [EMAIL PROTECTED]CSeq: 1 REGISTERFrom: sip:[EMAIL PROTECTED];tag=1654265612299Max-Forwards: 70To: sip:[EMAIL PROTECTED]User-Agent: SJphone/1.60.289a (SJ Labs) 10 headers, 0 linesUrgent handlerSip read:REGISTER sip:172.16.0.4 SIP/2.0Via: SIP/2.0/UDP62.0.0.1:49917;rport;branch=z9hG4bK0a03001043bac9d423f50004 Content-Length: 0Contact: sip:[EMAIL PROTECTED]:5060Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERFrom: sip:[EMAIL PROTECTED];tag=165427961757Max-Forwards: 70To: sip:[EMAIL PROTECTED]User-Agent: SJphone/1.60.289a (SJ Labs) Authorization: Digestusername=agallo,realm=asterisk,nonce=153ea438,uri=sip:172.16.0.4,response=9fc986f534bc3ac8ed8901df02cfa94e SIP/2.0 100 TryingVia: SIP/2.0/UDP62.0.0.1:49917;branch=z9hG4bK0a03001043bac9d423f50004;received=62.0.0.1;rport=5060From: sip:[EMAIL PROTECTED];tag=165427961757To: sip:[EMAIL PROTECTED];tag=as580bd78fCall-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED] Content-Length: 0to 62.0.0.1:5060Transmitting (NAT):SIP/2.0 403 ForbiddenVia: SIP/2.0/UDP62.0.0.1:49917;branch=z9hG4bK0a03001043bac9d423f50004;received= 62.0.0.1;rport=5060From: sip:[EMAIL PROTECTED];tag=165427961757To: sip:[EMAIL PROTECTED] ;tag=as580bd78fCall-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED]Content-Length: 0___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] integration with Meridian/Norstar ATA2
On Tuesday 03 January 2006 20:14, Darrick Hartman wrote: I'm attempting to use an asterisk box with a Digium TDM01B as voicemail for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're having problems where hangup is not always (but sometimes) detected. It's not detected probably 70% of the time or more. (The users transfer callers to an extension--caller then has to navigate a menu to get to the appropriate user). One solution you could use is to simply set an AbsoluteTimeout() before hitting VoiceMail, so that after 60 seconds it drops the call with prejudice. 60 seconds actually is quite a long time to store a message, and the odd time that it hangs up on a caller may be acceptable. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip-info: Asterisk record calls
Hi Tim! Wow, I didn't imagine that asterisk on different systems would use different date codes for the monitor filenames -- but aah isn't asterisk ;) My monitor filenames include the date and time, embedded as seconds since epoch iirc: [EMAIL PROTECTED] monitor]$ ll auto-1136394539-112-7476011-in.wav auto-1136394539-112-7476011-out.wav [EMAIL PROTECTED] monitor]$ ll auto-1136394539-112-7476011.wav So the 1136394539 part is seconds since epoch, 112 is who started the recording, 7476011 is where they were connected to when it happened. And, I suspect the auto- part is 'cause I used automon feature to do this? I haven't looked at asterisk code enough to see what filenames are created when. Thank you for the patch though. Now that I know many people are trying this stuff, I'll try to incorporate autodetection of filename style Moj *the file names I have are like below and I've decoded some of the sections - but not all of them *g1-20051205-215232-1133841147.211.WAV **g1- is this the zap channel 20051205- the date - should use this in the web page display instead 215232- time call was made I beleive 113 I dont know what the first 3 digits of this section are - 113 is source channel making the call they aren't part of the phone number 3841147.the 7 digit phone number 211.not sure on this either **and I've noticed If I call in a broadcast conference call and I don't say anything then it doesn't make a g1 file it leaves the out file like below and **these show in the index.php also which is good.* *** OUT207-20051106-184952-1131324592.448.WAV **OUT207- which direction and the extension number makeing the call 20051106- date 184952- time 113 ? 113 in this section matches all the files in my monitor directory 1324592.7 digit phone number 448.? * as I noted above - it looks to me like using the time and date in the file name would be more accurate - at least on my system. I'll have to brush up my php and fiddle with it. :) * * ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Resolving timing issues with dual PRIs in a TE411P
On Tuesday 03 January 2006 18:34, Casey Boone wrote: you could try setting the * box to pull timing from each pri connected to it and set the nortel to be a master for that circuit and see if that helps any That's actually what he has right now, and that's not such a good idea. Digium PRI cards can only take ONE clock and that is the clock for the entire card. He needs to build a proper spanning tree, but it appears that he has done this. The Allstream PRI (only four B channels?!) is the primary timing source, and if the Allstream PRI is down, the card will take timing from the Meridian. I would suggest changing the Meridian clocking to '0' (never sync to this span) and ensure that the Meridian is indeed set to recover clock from the line. (This is the default with Meridians). Short of that, I'd be looking at the system itself, looking for timing errors or overutilized CPU to try and track down the source of hte problem. Generally speaking you do not get echo or shitty call quality from frame slips; you get buzzing and HDLC errors and D channel restarts. -A. He should be slaving to the Allstream PRI ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE411P in a HP DL360 - which BIOS settings work?
Greetings, For a couple of months, we ran a pilot implementation of Asterisk with a TE110P on a HP NetServer LP1000r with perfect audio quality and no echo on a T1. Encouraged by this, we purchased two identical HP DL360 servers with TE411P cards for a production installation of Asterisk. We reproduced, as close as we were able, the configuration that we had on the pilot server. This is what we have now on our DL360s (twin 3.4GHz Xeons and 4GB of memory): - Latest stable build of asterisk, zaptel and libpri (1.2.1) - APIC turned off - Hyperthreading disabled - USB disabled - Physical and virtual COM ports disabled - TE411P on its own on IRQ4 We have also tried APIC and hyperthreading on with no perceptible difference. We have problems with echo, scratchy audio, pops, clicks and drop outs that we cannot seem to make go away. We have had kernel panics when downing network interface cards, and, on one of the servers, kernel messages that say Disabling IRQ #11 when we modprobe wct4xxp. Is there anyone out there who has gotten this combination of hardware working satisfactorily and would like to share their BIOS and IRQ settings and anything else they did to get it working? I know it must work - the DL360 is one of the two supported servers for ABE. We have emailed Digium support and are waiting for a response from them, but I wanted to poll the list and see if there was anyone with a working implementation out there. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Having major issues with TDM2400
On Wednesday 04 January 2006 01:34, Kerry Garrison wrote: Not at all, I am right with you. I am listening to what Digium is saying and letting them spin their resources on it. They say they have it working, Who at Digium is saying that POTS inband progress detection will definitely work and that they will make it work no matter what? Nobody at Digium that I know would ever say anything like that, because inband progress detection is a fool's errand in this day and age. It *can* work, but it's difficult, not guaranteed and is prone to misdetection. The second you *rely* on it, as it appears you are trying to do, that is when it will jump up and bite you on the ass. This is one of the biggest reasons why anyone doing this professionally uses a PRI. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] M0n0Wall traffic shaping rules
On Wed, 2006-01-04 at 11:59 -0500, Mark Phillips wrote: Anyone got any VoIP traffic shaping rules for m0n0wall that they could let me look at please? Running m0n0wall-1.21 now, I used the wizard to set the base queues/pipes/rules then added two more rules: If Dir Proto Src Dst TargetDescription --- --- - --- - --- WAN - UDP pbx:4569 *:4569 m_High Priority #1 Upload IPX VoIP WAN - UDP *:4569 pbx:4569 m_High Priority Download IPC VoIP I have this setup at two sites that use an IAX ITSP and also connect directly to each other. Seems to work fine but I'm not really sure how to actually prove that it's 100% correct. I'd love to hear if you get anything better. I'm not using SIP externally but I'd assume the same rules would work with 5060 for the port. HTH, Paul -- Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA -- On site at GDOT's W.Annex, 404-463-2860 x199 -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Driver for channel SIP/210-7450' does not support indication 3, emulating it
All of my phones are Cisco 7960's. Each one of them occasionally show up in the logs with the follow message: messages:Oct 21 09:35:43 DEBUG[25423] channel.c: Driver for channel 'SIP/210-7450' does not support indication 3, emulating it What does this mean and how do I fix it? I am using asterisk-1.2.1 with Cisco SIP version 7.4. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AMP: Losing backslash characters in config files
Steve Langstaff wrote: I've just started using AMP and found that I have a problem with escaped characters in config files. In particular, I have a custom config item that needs a semicolon in... SetVar(_ALERT_INFO=info=auto-answer;delay=1) To get the part of the line after the ; to be accepted by Asterisk as a non-comment it needs to be escaped with a backslash, but I have found that I need to put two backslashes in to get this through AMP. Subsequent to this, when AMP displays the page it has stripped one of the backslashes. The outcome of this is that if you view and then update a working config file it breaks. Try mailing the AMP mailing list as this is not a problem with Asterisk and this is the _Asterisk_ users list :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] M0n0Wall traffic shaping rules
Paul Dugas wrote: On Wed, 2006-01-04 at 11:59 -0500, Mark Phillips wrote: Anyone got any VoIP traffic shaping rules for m0n0wall that they could let me look at please? Running m0n0wall-1.21 now, I used the wizard to set the base queues/pipes/rules then added two more rules: I don't use m0n0wall, but wouldn't it be better just to shape based on a Type Of Service and then set the TOS flags in iax.conf and sip.conf accordingly? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] integration with Meridian/Norstar ATA2
Andrew Kohlsmith wrote: On Tuesday 03 January 2006 20:14, Darrick Hartman wrote: I'm attempting to use an asterisk box with a Digium TDM01B as voicemail for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're having problems where hangup is not always (but sometimes) detected. It's not detected probably 70% of the time or more. (The users transfer callers to an extension--caller then has to navigate a menu to get to the appropriate user). One solution you could use is to simply set an AbsoluteTimeout() before hitting VoiceMail, so that after 60 seconds it drops the call with prejudice. 60 seconds actually is quite a long time to store a message, and the odd time that it hangs up on a caller may be acceptable. Forgot to mention that the boss likes to call in and dictate to the secretary on the voicemail system. (so the maximum message length was set to 3600 seconds) If I'm reading correctly AbsoluteTimeout() would limit the length of the call to whatever value is set. If that's 60 seconds, the max length of the vm message would essentially be 60 seconds. That won't work for this solution. I'll try silence detection in the voicemail.conf file because so far, that seems like the best solution. Another option would be to force the boss to call in on the analog line that we have dedicated to Asterisk for any dictations. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Driver for channel SIP/210-7450' does not support indication 3, emulating it
Jeremy Koski wrote: messages:Oct 21 09:35:43 DEBUG[25423] channel.c: Driver for channel 'SIP/210-7450' does not support indication 3, emulating it What does this mean and how do I fix it? I am using asterisk-1.2.1 with What makes you think there is something to fix? This is a DEBUG message. If you don't know what DEBUG messages are for, don't turn them on. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NOTICE[3589]: frame.c:128 __ast_smoother_feed: Dropping extra fr ame of G.729 since we already have a VAD frame at the end
Hello. Im using Asterisk like IVR card application. It works very well in h323 and SIP, but when the IVR generate a call in SIP it show: Jan 4 15:39:32 NOTICE[3589]: frame.c:128 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end As I see, this is a problem originated by a equipment with VAD activated. But we disable VAD in all our equipment. Someone have ani clue about it? Regards. jsalas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial(Console/dsp) and option g doesnt appear to work
I have a case where I need the option g to continue execute after the hangup (I'm using 1.2.1) and I have the following in my extensions: exten = 309,1,System(echo /tmp/file) exten = 309,2,Dial(Console/dsp,,g) exten = 309,3,System(rm -f /tmp/file) exten = 309,4,Hangup However, after the hangup priority 3 is not executed. Does 'g' not work with console/dsp or do I have something wrong. THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] integration with Meridian/Norstar ATA2
Darrick Hartman wrote: Andrew Kohlsmith wrote: On Tuesday 03 January 2006 20:14, Darrick Hartman wrote: I'm attempting to use an asterisk box with a Digium TDM01B as voicemail for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're having problems where hangup is not always (but sometimes) detected. It's not detected probably 70% of the time or more. (The users transfer callers to an extension--caller then has to navigate a menu to get to the appropriate user). One solution you could use is to simply set an AbsoluteTimeout() before hitting VoiceMail, so that after 60 seconds it drops the call with prejudice. 60 seconds actually is quite a long time to store a message, and the odd time that it hangs up on a caller may be acceptable. Forgot to mention that the boss likes to call in and dictate to the secretary on the voicemail system. (so the maximum message length was set to 3600 seconds) If I'm reading correctly AbsoluteTimeout() would limit the length of the call to whatever value is set. If that's 60 seconds, the max length of the vm message would essentially be 60 seconds. That won't work for this solution. I'll try silence detection in the voicemail.conf file because so far, that seems like the best solution. Another option would be to force the boss to call in on the analog line that we have dedicated to Asterisk for any dictations. Darrick Probably I'm wrong, but it is not better to use an TDM10B (1) FXS connected to an Meridian CO port? Jorge ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
