Re: [Asterisk-Users] ztdummy
On Thu, Jan 26, 2006 at 03:10:09PM -0600, Mike Hammett wrote: I'm running a VPS and I need to pass the device drivers from the host OS to the VPS. What files do I need to pass through for ztdummy to work? I'm assuming they're in /dev/zap, but I'm not sure which ones are needed. ztdummy (of kernel 2.6) should not require anything from the host. However are you sure you can use different kernels for the host and the guest with your VPS? It does generate a load of 1000 interrupts per second. This means tha tyou always need CPU time. And a timely response of it. -- 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
Re: [Asterisk-Users] Max concurrent calls
There is no such thing as a hard limit in asterisk. (Except for zap channels, those are limited to 256 iirc). With iax you can go higher, but the limit might be lower than 256 if you are doing a lot of transcoding. The limit depends on what exactly the server has to do with your call, and how fast your server is. Zoa Andrew Nowrot wrote: Hi, Does anyone know what is the amount of max concurrent calls that can be made in one Asterisk box? I heard that it is 256 and it doesn't depend on how good your machine is. It is the program constraint. What can I do when I need to have more calls than that. I read about connecting Asterisk boxes with IAX. Is it a good solution? Does anyone have other proposals? Cheers Andrew ___ --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] Asterisk authorization
Do anyone know how to setup asterisk to authenticate the user through IP rather than username and password? I know most carriers will do that but smaller end user providers will not do. Sam ___ --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] Dynamically disabling echo cancellation (Zap).
Hi! For reasons that I won't bore people with, I'd like to disable echo cancellation on-the-fly, depending on which DID a call came in on. I've seen things like spandsp disable EC for faxes, so I know it's possible. Any idea where to start looking? (I assume I'll have to make a helper application of some sort to be called externally, and that's fine.) If you are using bristuff you could use the m option Dial(Zap/g1m/Number) in order to get rid of rxgain/txgain and AFAIK echo cancellation. This has been built into bristuff esp. for faxes etc. Regards, JP ___ --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] Max concurrent calls
Andrew Nowrot a écrit : Hi, Does anyone know what is the amount of max concurrent calls that can be made in one Asterisk box? I heard that it is 256 and it doesn't depend on how good your machine is. It is the program constraint. I wasn't aware of such limit and I seriously doubt it. Where are you pulling this number from? (other than the obvious traditional 2^8)? -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] paging agi
Hi
Some petty notes notes regarding the perl:
On Thu, Jan 26, 2006 at 11:23:27PM -0800, Jeremy wrote:
1. you didn't use strict and -w. Debugging will be a whole lot tougher
2. Consider using the nagging -T (taint mode), to explicitly know when
you trust the input.
3. Consider the latency this cases. Note that this script is run at the
startup of every call. Let's see what it does:
#
#Check for Telnet
if (eval require Net::Telnet;) {
use Net::Telnet;
} else {
print VERBOSE \Net::Telnet NOT INSTALLED - this is required\
0\n;
exit 0;
}
Just use:
use Net::Telnet;
This will generate a compile-time (perl -c) check which is easy to test
automatically.
BTW: why not use an existing perl module to talk to
# You need to configure this: Your manager API username and password. This
# is the information from /etc/asterisk/manager.conf. You need something
like
# this in it:
# [admin]
# secret = amp111
# deny=0.0.0.0/0.0.0.0
# permit=127.0.0.0/255.0.0.0
# read = system,call,log,verbose,command,agent,user
# write = system,call,log,verbose,command,agent,user
Now, why generate entries to talk to the the manager interface on
localhost with full the permissions? Are all of those permissions
necessary?
Wouldn't it make more sense to send commands directly to the unix domain
socket of Asterisk? Basically you connect there and send separate
commands in separate writes.
# IF that's what you have in your conf file, this is what you should have
here:
my $mgruser = admin;
my $mgrpass = amp111;
my $mgrport = 5038;
# set our array of phones that we will NOT be paging
@bypass = @ARGV;
# Get our needed info for idle sip phones
@sips = grep(/^\s+\d+\s.*/, `asterisk -rx show hints`);
ah, but you do use it, indirectly. Why not get that information from the
manager interface?
# Now check each sip phone to see if it's in use and also
# against our exclude list. If it passes both, it's
# added to our array of calls to make
foreach $sipline (@sips) {
my ($junk0, $exten, $junk1, $chan, $state, $junk2) = split(/ +/,
$sipline,6);
unless (($state ne State:Idle) || (grep(/$exten/i, @bypass))) {
push (@mypage, $exten);
}
}
$page= join (SIP/, @mypage);
print $page;{
# Open connection to AGI and set Global Variable PAGE_GROUP
# to our completed array of sip phones
my $tn = new Net::Telnet ( Port = $mgrport,
Prompt = '/.*[\$%#] $/',
Output_record_separator = '',
Input_Log= /tmp/input.log,
Output_Log= /tmp/output.log,
Errmode= 'return', );
$tn-open(127.0.0.1);
$tn-waitfor('/0\n$/');
$tn-print(Action: Login\n);
$tn-print(Username: $mgruser\n);
$tn-print(Secret: $mgrpass\n);
$tn-print(Events: off\n\n);
my ($pm, $m) = $tn-waitfor('/Authentication (.+)\n\n/');
if ($m =~ /Authentication failed/) {
print VERBOSE \Incorrect MGRUSER or MGRPASS - unable to connect to
manager interface\ 0\n;
exit;
}
$tn-print(Action: Setvar\n);
$tn-print(Variable: PAGE_GROUP\n);
$tn-print(Value: $page\n\n);
$tn-print(Action: Logoff\n\n);
$tn-close;
}
Then add the following to your dialplan:
exten = *61,1,Set(TIMEOUT(absolute) = 15)
exten = *61,n,AGI(page.agi|arg|arg)
exten = *61,n,SetCallerID(Page:${CALLERIDNAME} ${CALLERIDNUM})
exten = *61,n,Set(_ALERT_INFO=Ring Answer)
exten = *61,n,SIPAddHeader(Call-Info: answer-after=0)
exten = *61,n,Page(SIP/${PAGE_GROUP})
exten = *61,n,Hangup()
I really don't see why you need to connect to the manager just to set a
global variable. You can easily do that from the dialplan or from the
AGI itself. Or am I missing anything?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
___
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RE: [Asterisk-Users] Chan_capi on builds 79558320 strangeness
/etc/init.d/asterisk stop Stopping Asterisk PBX: . censys:/usr/src/asterisk-8632# cd .. censys:/usr/src# asterisk -vc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk SVN-trunk-r8620, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: CLIP [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found == This box has 1 capi controller(s). -- CAPI/contr1 supports DTMF -- CAPI/contr1 supports echo cancellation -- CAPI/contr1 supports line interconnect -- CAPI/contr1 supports supplementary services supplementary services : 0x010f HOLD/RETRIEVE TERMINAL PORTABILITY ECT 3PTY MWI == Reading config for ISDNL1 -- capi_pvt ISDNL1-pseudo-D (5912211,capi-in-5912211,0,2) (1,4,64) -- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64) -- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64) == Reading config for ISDNL2 -- capi_pvt ISDNL2-pseudo-D (6930821,capi-in-6930821,0,2) (0,0,64) -- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64) -- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64) -- listening on contr1 CIPmask = 0x1fff03ff == Registered channel type 'CAPI' (Common ISDN API Driver (cm-0.6.3) ) == Registered application 'capiCommand' == Registered custom function VANITYNUMBER CLIP Asterisk Ready. *CLI capi debug CAPI Debugging Enabled *CLI -- Saved useragent PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 for peer 364 -- Executing Set(SIP/366-11b2, CALLFILENAME=/var/spool/asterisk/monitor/outgoing/9145912211/Out-200601 18-030458-9145912211_ADCOM Office_19145912211) in new stack -- Executing SetCallerID(SIP/366-11b2, 9145912211) in new stack -- Executing Monitor(SIP/366-11b2, wav|/var/spool/asterisk/monitor/outgoing/9145912211/Out-20060118-030458 -9145912211_ADCOM Office_19145912211) in new stack -- Executing Dial(SIP/366-11b2, IAX2/ll/19145912211) in new stack -- Called ll/19145912211 -- Call accepted by 208.139.204.232 (format ulaw) -- Format for call is ulaw -- IAX2/teliaxcsi-8 is making progress passing it to SIP/366-11b2 -- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm Callctrl/1.5 MxSF/v3.2.6.26 for peer 347 -- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm Callctrl/1.5 MxSF/v3.2.6.26 for peer 345 -- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm Callctrl/1.5 MxSF/v3.2.6.26 for peer 361 -- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm Callctrl/1.5 MxSF/v3.2.6.26 for peer 363 -- Hungup 'IAX2/teliaxcsi-8' == Spawn extension (cisco-teliaxoutcsi, 19145912211, 4) exited non-zero on 'SIP/366-11b2' -- Saved useragent PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 for peer 365 -- Saved useragent PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 for peer 330 -- Executing Set(SIP/366-5e8d, CALLFILENAME=/var/spool/asterisk/monitor/outgoing/9145912211/Out-200601 18-030510-9145912211_ADCOM Office_19145912211) in new stack -- Executing SetCallerID(SIP/366-5e8d, 9145912211) in new stack -- Executing Monitor(SIP/366-5e8d, wav|/var/spool/asterisk/monitor/outgoing/9145912211/Out-20060118-030510 -9145912211_ADCOM Office_19145912211) in new stack -- Executing Dial(SIP/366-5e8d, IAX2/[EMAIL PROTECTED]/19145912211) in new stack -- Called [EMAIL PROTECTED]/19145912211 -- Call accepted by 208.139.204.232 (format ulaw) -- Format for call is ulaw Jan 17 22:05:11 WARNING[8571]: chan_iax2.c:7525 socket_read: Received mini frame before first full voice frame -- IAX2/teliaxcsi-9 is making progress passing it to SIP/366-5e8d CONNECT_IND ID=001 #0x0001 LEN=0050 Controller/PLCI/NCCI= 0x201 CIPValue= 0x1 CalledPartyNumber = c15912211 CallingPartyNumber = 21 819145912211 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a2 LLC = default HLC = default AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default -- CONNECT_IND
[Asterisk-Users] ASTPP
Hi, Has anyone implemented astpp? I'm configuring one right now and I have a problem on the pricelist. I followed the steps here http://www.astpp.org/index.php?n=ASTPP.Installation and created tables using http://www.astpp.org/index.php?n=ASTPP.Structure, but i didn't see there a query on creating pricelist table, can anyone help me on this please? Thank You Regards, Ronald ___ --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] Max concurrent calls
Hi,Yeah, I think it was all about thew zap channelsBut what opportunities I have when I need to connect two or more Asterisk boxes. IAX, SIP or what?What is most efficient.CheersAndrew ___ --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] Offtopic: Auto provioning Snom 360
Hello list, I've got a problem provisioning my snom 360's in the office (about 20 of them). I'm using DHCP option 66/67 to set the provisioning URL but the phone won't connect to it to retrieve it's configuration. We are using a Cisco Catalyst Epress 500 to power the phones (poe), however if i power hem via the adapter and hook them up to a hub instead of the switch the phone does retrieve it's config but we would like to use the poe to power them. Anyone else using a Catalyst Express 500 with snom's ? Kind regards, Erik Versaevel ___ --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] Announcement: Snom 360 with integrated XML Objects
I would like to add that I did have at one point problems figuring out 4.0 and there were no problems downgrading. Also I made a special email account @mydomain for SNOM liscence's. This helps if at a later dat you need to re-enter it again. Regards, Dovid --- Christian Stredicke [EMAIL PROTECTED] wrote: As far as the licence is concerned that is something that we introduced in the 4.0 image and this is not against our customers (which would be stupid). It shall protect us from clones. The jump to the 5.0 is not about this licensing stuff, we just changed the ramdisk and freed up more memory. I know this is not very pleasant, and we cross fingers that this is the last time we have to do something like this. But running out of memory is also not very pleasant! Especially when new cool features ask for more memory! CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Thursday, January 26, 2006 2:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Announcement: Snom 360 with integrated XML Objects that is *very* cool. However, I am somewhat concerned about being forced to license the firmware (even if it is free) can you comment for the list the rationale behind forcing a license and how this might affect Snom users who, say, want to DOWN grade their firmware? ps is there a timetable for supported, formally released firmware version? -Original Message- From: Hirosh Dabui [mailto:[EMAIL PROTECTED] Sent: Thursday, January 26, 2006 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Announcement: Snom 360 with integrated XML Objects -BEGIN PGP SIGNED MESSAGE- Hash: RIPEMD160 Dear user, the new snom 360 is able to use services from standard web servers. Users can deploy customized client services with snom 360 and interact with other users via the keypad. The snom 360 will use HTTP protocol from standard web servers, like Apache. Typical services are: ~ 1. To-do lists ~ 2. Stock Information ~ 3. Weather ~ 4. Provisioning ~ 5. Agenda ~ 6. Telephone directory For further information go to http://snom.com/wiki/index.php/Xmlobjects Note: *That is a pre-release, probably the software is still unstable* Best regards, Hirosh Dabui - -- snom technology AG Dipl.-Ing. Hirosh Dabui PGP Key-ID: 0x30A34758 mailto:[EMAIL PROTECTED] http://snom.com -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (GNU/Linux) iD8DBQFD2Q6YAO47/DCjR1gRA6REAJ4iSyot8OhFVDt0/C2I7KFoRCP18ACeNGau FCXMUdN9loiwy948EO8th9U= =Qntp -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk authorization
Hello Sam, use host=IP_ADDRESS when defining user in sip.conf regards, Umair Bari On 1/26/06, Sam Tam [EMAIL PROTECTED] wrote: Do anyone know how to setup asterisk to authenticate the user through IPrather than username and password? I know most carriers will do that but smaller end user providers will notdo.Sam___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Max concurrent calls
Hi, Where are you pulling this number from? (other than the obvious traditional 2^8)? That is not my imagination ;).Actually I talked with a guy who was one of the designers of Asterisk. He told me about this limitation but I don't know if he was talking about Zap channels only or in general. I will ask him asap. CheersAndrew ___ --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] Packeting multiple GSM frames in one IP packet - Help needed.
Hi, We have a task to reduce voice call bandwidth. IP+UDP+RTP are using 40 bytes per packet and for voice GSM FR 33 bytes. We are trying to reduce this bandwidth accommodating multiple GSM frames in one packet. If we want to use per packet 10 GSM frames how to do this using asterisk? Assume the sip client is able to split these packets in to individual GSM frames. Any help will be sincerely appreciated. Thanks Regards, Sheshu. ___ --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] Max concurrent calls
Andrew Nowrot a écrit : Hi, Yeah, I think it was all about thew zap channels But what opportunities I have when I need to connect two or more Asterisk boxes. IAX, SIP or what? What is most efficient. Your question doesn't make any sense. Tell us what you are trying to do and you might get meaningful replies. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] Max concurrent calls
Andrew Nowrot a écrit : Hi, Where are you pulling this number from? (other than the obvious traditional 2^8)? That is not my imagination ;). Actually I talked with a guy who was one of the designers of Asterisk. He told me about this limitation but I don't know if he was talking about Zap channels only or in general. I will ask him asap. It does sound like a typical case of urban legend, where Zap is limited to 256 channels becomes Asterisk is limited to 256 channels. Asterisk != Zap. Speaking of Zap, what happened to Digiums' DS3 card? Did it got hit by the 256 channel limit? :-) Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] DeadAGI and Hangup on channel
Hello, I'm trying to catch channel hangup in DeadAgi script. For example, A calls to DeadAgi script which connects (Dial) to B. After Dial command exits I need to identify where hangup came from: A or B. CHANNEL STATUS returns 6 (Line is Up) regardless of who hungup. In CLI show channels states that channel A to DeadAgi is UP even if A and B hungup. If A stays on the line after conversation with B (hangup from B), then DeadAgi would continue (with prompts and etc.), if A is off then DeadAgi should exit gracefully (not killed as Agi). Does anyone know how to do it? Thanks in advance. -- Grigoriy Puzankin ___ --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] No matching peer or user based on IP address
Hi all, I'm running Asterisk SVN-trunk-r8643M and face following problem: I'm trying to get incoming call from a provider and calls ended with a 404 error. On the INVITE I get Found no matching peer or user for IP address:5060 and then Looking for UserName in SIP default context (domain xxx.xxx.xxx.xxx). My question is why asterisk doesn't found my peer/user chapter? If I add an extension UserName,1,blablabla in my SIP default context, it's working. The provider has multiple IP address. Here is sip.conf and debug logs: [UserName] type=user ;tested with friend username=UserName fromuser=UserName fromdomain=ProviderDomain secret=MySecret context=from-provider host=sip.ProviderDomain.com insecure=port,invite;tested with very nat=no canreinvite=no disallow=all allow=alaw,ulaw ;g726 Jan 27 00:42:44 VERBOSE[16980] logger.c: --- (11 headers 8 lines)Jan 27 00:42:44 VERBOSE[16980] logger.c: --- (11 headers 8 lines)--- Jan 27 00:42:44 VERBOSE[16980] logger.c: Using INVITE request as basis request - [EMAIL PROTECTED] Jan 27 00:42:44 VERBOSE[16980] logger.c: Sending to xxx.xxx.xxx.xxx : 5060 (non-NAT) Jan 27 00:42:44 VERBOSE[16980] logger.c: Found no matching peer or user for 'xxx.xxx.xxx.xxx:5060' Jan 27 00:42:44 VERBOSE[16980] logger.c: Found RTP audio format 8 Jan 27 00:42:44 VERBOSE[16980] logger.c: Peer audio RTP is at port yyy.yyy.yyy.yyy:11274 Jan 27 00:42:44 VERBOSE[16980] logger.c: Found description format pcma Jan 27 00:42:44 VERBOSE[16980] logger.c: Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Jan 27 00:42:44 VERBOSE[16980] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Jan 27 00:42:44 VERBOSE[16980] logger.c: Looking for UserName in SIP default context (domain xxx.xxx.xxx.xxx) Jan 27 00:42:44 VERBOSE[16980] logger.c: Reliably Transmitting (no NAT) to xxx.xxx.xxx.xxx:5060: SIP/2.0 404 Not Found Thank's for any hint -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan_capi on builds 79558320 strangeness
This is not a problem of the ISDN line (or chan_capi), Asterisk is just not doing anything after -- Executing GotoIfTime(CAPI/ISDNL1/5912211-0,20:01-7:59|mon-sun|*|*?9) in new stack and without further commands (like Ringing(), Answer(), ...) the ISDN line timed out and disconnects. So either your dialplan is buggy, or Asterisk is not doing what you want. What should be done according your extensions.conf in that state ? Armin On Fri, 27 Jan 2006 [EMAIL PROTECTED] wrote: /etc/init.d/asterisk stop Stopping Asterisk PBX: . censys:/usr/src/asterisk-8632# cd .. censys:/usr/src# asterisk -vc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk SVN-trunk-r8620, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log Asterisk Dynamic Loader loading preload modules: CLIP [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found == This box has 1 capi controller(s). -- CAPI/contr1 supports DTMF -- CAPI/contr1 supports echo cancellation -- CAPI/contr1 supports line interconnect -- CAPI/contr1 supports supplementary services supplementary services : 0x010f HOLD/RETRIEVE TERMINAL PORTABILITY ECT 3PTY MWI == Reading config for ISDNL1 -- capi_pvt ISDNL1-pseudo-D (5912211,capi-in-5912211,0,2) (1,4,64) -- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64) -- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64) == Reading config for ISDNL2 -- capi_pvt ISDNL2-pseudo-D (6930821,capi-in-6930821,0,2) (0,0,64) -- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64) -- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64) -- listening on contr1 CIPmask = 0x1fff03ff == Registered channel type 'CAPI' (Common ISDN API Driver (cm-0.6.3) ) == Registered application 'capiCommand' == Registered custom function VANITYNUMBER CLIP Asterisk Ready. *CLI capi debug CAPI Debugging Enabled *CLI -- Saved useragent PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 for peer 364 -- Executing Set(SIP/366-11b2, CALLFILENAME=/var/spool/asterisk/monitor/outgoing/9145912211/Out-200601 18-030458-9145912211_ADCOM Office_19145912211) in new stack -- Executing SetCallerID(SIP/366-11b2, 9145912211) in new stack -- Executing Monitor(SIP/366-11b2, wav|/var/spool/asterisk/monitor/outgoing/9145912211/Out-20060118-030458 -9145912211_ADCOM Office_19145912211) in new stack -- Executing Dial(SIP/366-11b2, IAX2/ll/19145912211) in new stack -- Called ll/19145912211 -- Call accepted by 208.139.204.232 (format ulaw) -- Format for call is ulaw -- IAX2/teliaxcsi-8 is making progress passing it to SIP/366-11b2 -- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm Callctrl/1.5 MxSF/v3.2.6.26 for peer 347 -- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm Callctrl/1.5 MxSF/v3.2.6.26 for peer 345 -- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm Callctrl/1.5 MxSF/v3.2.6.26 for peer 361 -- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm Callctrl/1.5 MxSF/v3.2.6.26 for peer 363 -- Hungup 'IAX2/teliaxcsi-8' == Spawn extension (cisco-teliaxoutcsi, 19145912211, 4) exited non-zero on 'SIP/366-11b2' -- Saved useragent PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 for peer 365 -- Saved useragent PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 for peer 330 -- Executing Set(SIP/366-5e8d, CALLFILENAME=/var/spool/asterisk/monitor/outgoing/9145912211/Out-200601 18-030510-9145912211_ADCOM Office_19145912211) in new stack -- Executing SetCallerID(SIP/366-5e8d, 9145912211) in new stack -- Executing Monitor(SIP/366-5e8d, wav|/var/spool/asterisk/monitor/outgoing/9145912211/Out-20060118-030510 -9145912211_ADCOM Office_19145912211) in new stack -- Executing Dial(SIP/366-5e8d, IAX2/[EMAIL PROTECTED]/19145912211) in new stack -- Called [EMAIL PROTECTED]/19145912211 -- Call accepted by 208.139.204.232 (format ulaw) -- Format for call is ulaw Jan 17 22:05:11 WARNING[8571]: chan_iax2.c:7525 socket_read: Received mini frame before first full voice frame -- IAX2/teliaxcsi-9 is making progress passing it to SIP/366-5e8d CONNECT_IND ID=001 #0x0001
[Asterisk-Users] ODBC Problem with voicemail.
I've installed the last released asterisk 1.2.2 on my own HLFS system with a 2.6.14.3 kernel. I've also a 2 FXO/ 1 FXS digium card on it. Every thing is working correctly. For ODBC, I'm using UnixODBC with pgsql. The voice messages are correctly written to the database and also their number is correctly reported by VoicemailMain dialogue. However, after reading the time/day of the new message, the system hangup without playing the message and automatically reports that it couldn't open /var/spool/.../msg.WAV, from the file.c (ast_filehelper) and Unable to open /var/.../msg from the file.c (ast_streamfile) methods. After some dubbeguing (I've a sptripped system, I don't have access to all symbols), it seems like the temporary files are not created at all and the open method (in retieve_file method) return always -1 which prevent from creating the file of the message. The directory is writable for the asterisk user but not for the group. Even the description file of the message is created correctly as I've seen. My question is: does asterisk create the file even if the message is empty or not? If it doesn't so the problem could come from the ODBC part as asterisk is not able to read the data even if message is correctly recorded in the database. The message is reported as read and thus becomes old. I checked that from the code and this happens from the playing_message method as expected. Is there any people in the list who encoutered similar problems? Many thanks. Omar. ___ --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] ASTPP
Ronald Ramos wrote: Hi, Has anyone implemented astpp? I'm configuring one right now and I have a problem on the pricelist. I followed the steps here http://www.astpp.org/index.php?n=ASTPP.Installation and created tables using http://www.astpp.org/index.php?n=ASTPP.Structure, but i didn't see there a query on creating pricelist table, can anyone help me on this please? Thank You Regards, Ronald Under Rates click on - Pricelists then Add... -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Max concurrent calls
Hi, It does sound like a typical case of urban legend, where Zap is limited to 256 channels becomes Asterisk is limited to 256 channels. Asterisk!= Zap.I've never said that Asterisk is limited to 256 channels. I only asked a question. That is the main reason of this list isn't it? But leave the limitation thing :).I need to connect two (or more) asterisk boxes. They will exchange a lot calls. What is the best approach? Which protocol should I use IAX or SIP or what? I never did that so first I want to ask people who have some experience. Thanks in advanceCheers ___ --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] Max concurrent calls
I need to connect two (or more) asterisk boxes. They will exchange a lot calls. What is the best approach? Which protocol should I use IAX or SIP or what? I never did that so first I want to ask people who have some experience. If you're connecting Asterisk boxes between each other, it would make sense to use IAX as it's Asterisk's 'native' protocol. Things which weight in favor of SIP. [1]. - It's an industry standard, so you can interoperate with many other SIP compliant systems. - Signaling and RTP are separate, so you have a central signaling / routing / billing server, for example using SER. Things which weight in favor of IAX. [2] [3] - Works very well in NATed / Firewalled environments since it uses only one UDP port and needs only one of both parties to be accessible. So I would say that if you are having only asterisk boxes to interconnect, IAX is totally the way to go. It will be a lot easier for you to setup, especially firewall / NAT wise. If your Asterisk boxes are fitted with timing devices (such as TDM4XX cards) you might even want to try IAX trunking to save some bandwith. Although personally, if bandwith is not a problem, I would leave trunking out of the equation. If you have more than 2 * boxes, you might want to try Dundi [4] to have some kind of shared dialplan between the boxes. Hope this helps. Cheers, Jean-Michel. [1] http://www.voip-info.org/wiki/view/SIP [2] http://www.voip-info.org/wiki-IAX [3] http://www.voip-info.org/wiki-IAX+versus+SIP (read the comments also!) [4] http://www.dundi.com/ -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] Is Voxee down?
HiI had same problems yesterday but ts fine now,DanOn 27/01/06, Mark Adams [EMAIL PROTECTED] wrote: I would expect a reply in about 4-5 days … From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Angelito Manansala Sent: Thursday, January 26, 2006 8:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is Voxee down? can you send calls? On 1/27/06, Guillermo Salas M [EMAIL PROTECTED] wrote: Con fecha 26/1/2006, Angelito Manansala [EMAIL PROTECTED] escribió: Hi Guys, We cant send calls and register to voxee server. however we already send support ticket and waiting for their reply. anybody experience this also on voxee? The same issue. Can not register. -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +63 917 542 5807 DID: (+63) 44 7906770 US DID: +1 619 399 0128 msn: [EMAIL PROTECTED] skype: bulcrac ___ --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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +63 917 542 5807 DID: (+63) 44 7906770 US DID: +1 619 399 0128 msn: [EMAIL PROTECTED] skype: bulcrack ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] paging agi
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: 27 January 2006 08:21
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] paging agi
Hi
Some petty notes notes regarding the perl:
SNIPPED
I really don't see why you need to connect to the manager just to set
a
global variable. You can easily do that from the dialplan or from the
AGI itself. Or am I missing anything?
Hi all,
I'm not sure if this helps but I had a bit of a bash at fixing some
issues with the new Page() function.
The problem was the phone that initiates the Page was getting called
itself and I think because the Snom360 had a bit of a race condition it
wasn't returning 486 Busy Here as it should of and thus causing a
feedback loop that killed the phone and required a reboot to fix.
My solution was the following (advice is more than welcome as this is
the first EVER bash script I have written and maybe GROUP function would
have been an option):
Dialplan:
[macro-page]
exten = s,1,SIPAddHeader(Call-Info:
sip:192.168.10.16\;answer-after=0)
;;;Strip out calling channel
exten = s,n,Cut(caller=CHANNEL,-,1)
;exten = s,n,Cut(caller=caller,/,2)
exten = s,n,AGI(strremove|${ARG1}|${caller})
exten = s,n,NoOp(dialstring: ${dialstring})
;exten = s,n,Cut(dialstr=ARG1,caller,1)
;exten = s,n,NoOp(dialstr: ${dialstr})
exten = s,n,Page(${dialstring})
/var/lib/asterisk/agi-bin/strremove:
#!/bin/sh
dialarg=` echo $1 | sed -e 's///g'`
channelarg=` echo $2 | sed -e 's///g'`
echo arg: $1
echo arg: $unqarg
OUT=test1 $string test2
OUT =${dialarg//$channelarg/}
echo SET VARIABLE dialstring $OUT
(I have tidied up the script a little to remove old debug but it should
still work fine.)
Notes:
- The AGI script finds and removes a string from the target string.
- The macro takes ARG1 as input which is the list of devices you wish to
use as intercom phones:
exten = 200,1,Macro(page,SIP/snom1SIP/snom2SIP/snom3) ;Pager /
Intercom
I still have issues with intercom breaking the odd snom that is
currently on the phone so I dare say that Jeremy's script will fix that.
However it would seem to me that the page function should be doing this
itself.
I hope this is of use to someone.
Alex
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[Asterisk-Users] WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38
Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo ___ --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] ATA's ???
Hi, I'm currently in the process of building Asterisk for our new office and have hit a snag. We need two internal Analog lines for a modem and fax machine. Am I right in thinking I can use two ATA's, one on each piece of equipment which will then talk to Asterisk and route via our ISDN30? If the above is corrent could you recommend a good model? Thanks in advance. Phil. ___ --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] Max concurrent calls
HiIn my environment I have to connect 6 * boxes with each other so IAX is probably the best solutionThanksCheers ___ --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] Outgoing FXO and CDR
Hi all, When I do outgoing calls via my FXO card (TDM400, analog line), they get always marked ANSWERED in my CDR. I guess it is not that easy for fxo to determine if there is actually a call or just ringing. But anyway, is there a way to get this working right? Thanks in advance, Henry ___ --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] Interconnectiong two Asterisk boxes [was: Max concurrent calls]
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Friday, January 27, 2006 10:49 AM If you're connecting Asterisk boxes between each other, it would make sense to use IAX as it's Asterisk's 'native' protocol. ... So I would say that if you are having only asterisk boxes to interconnect, IAX is totally the way to go. It will be a lot easier for you to setup, especially firewall / NAT wise. OK. If your Asterisk boxes are fitted with timing devices (such as TDM4XX cards) you might even want to try IAX trunking to save some bandwith. Although personally, if bandwith is not a problem, I would leave trunking out of the equation. Why do you say this? If you have more than 2 * boxes, you might want to try Dundi [4] to have some kind of shared dialplan between the boxes. Why 'more than 2'? I'm planning to connect two Asterisk boxes sharing their dialplan. I'd like to know the best practice to do it: - 'switch' statement - DUNDI - other I don't know DUNDI at all. Thanks Mimmus ___ --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] Spa3k and ISDN
Hello all, I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal analogue lines. The same number is assigned to these lines. These lines are connected to 2 spa3k registered to my asterisk box. When calls arrive, TR1 try to pass call to the first spa. If spa not takes the call immediately then try to pass to the other spa. The only configuration I found works is to put the parameter 'PSTN Answer Delay' to 0 in each spa. The problem is Call CID. I suppose the problem is that Asterisk not sees the CID because the spa takes several seconds to know. In the Spa status page appears the CID but never in the asterisk box or extensions. If somebody can help me it would be appreciate, Regards, Manuel ___ --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] pb with callerid
Since I passed from version 1.0 to the 1.2.3. I have Pb with the
callerid. If somebody call with presentation of the number all is well.
If somebody make call in masked number, i couldn't send a callerid to
the phone.
It is in a call center and i use the callerid to present the name of the
number called to the operator.
Before that went. To identify the sda, I use the assignment of the
callerid according to the sda called.
Thank's for your help
Here what I do:
exten = 8489,1,AGI(test.php)
exten = 8489,n,Set(CALLERID(all)=${NOM_CLIENT} 123456789)
exten = 8489,n,AGI(test.php)
exten = 8489,n,Dial(SIP/7297,,T)
Presentation of number
-- Accepting call from '611134024' to '8489' on channel 0/8, span 1
-- Executing AGI(Zap/8-1, test.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
test.php: agi_request = test.php
test.php: agi_channel = Zap/8-1
test.php: agi_language = fr
test.php: agi_type = Zap
test.php: agi_callerid = 611134024
test.php: agi_calleridname = unknown
test.php: agi_dnid = 8489
test.php: agi_uniqueid = 1138355705.1362
test.php: agi_extension = 8489
test.php: agi_priority = 1
test.php: 2006-01-27 10:55:05
-- AGI Script Executing Application: (SetGlobalVar) Options:
(NOM_CLIENT=DSOFT)
== Setting global variable 'NOM_CLIENT' to 'DSOFT'
test.php: FIN
-- AGI Script test.php completed, returning 0
-- Executing Set(Zap/8-1, CALLERID(all)=DSOFT 123456789) in
new stack
-- Executing AGI(Zap/8-1, test.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/test.php
test.php: agi_request = test.php
test.php: agi_channel = Zap/8-1
test.php: agi_language = fr
test.php: agi_type = Zap
test.php: agi_callerid = 123456789
test.php: agi_calleridname = DSOFT
test.php: agi_dnid = 8489
test.php: agi_uniqueid = 1138355705.1362
test.php: agi_extension = 8489
test.php: agi_priority = 3
test.php: 2006-01-27 10:55:05
-- AGI Script Executing Application: (SetGlobalVar) Options:
(NOM_CLIENT=DSOFT)
== Setting global variable 'NOM_CLIENT' to 'DSOFT'
test.php: FIN
-- AGI Script test.php completed, returning 0
-- Executing Dial(Zap/8-1, SIP/7297||T) in new stack
We're at 10.101.51.252 port 14324
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 10.101.51.248:2051:
INVITE sip:[EMAIL PROTECTED]:2051;line=ld48ci1w SIP/2.0
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport
From: DSOFT sip:[EMAIL PROTECTED];tag=as417ffda1
To: sip:[EMAIL PROTECTED]:2051;line=ld48ci1w
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 27 Jan 2006 09:55:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 25657 25657 IN IP4 10.101.51.252
s=session
c=IN IP4 10.101.51.252
t=0 0
m=audio 14324 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 7297
IPBX-TEST*CLI
-- SIP read from 10.101.51.248:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport=5060
From: DSOFT sip:[EMAIL PROTECTED];tag=as417ffda1
To: sip:[EMAIL PROTECTED]:2051;line=ld48ci1w;tag=okikki5eaw
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Contact: sip:[EMAIL PROTECTED]:2051;line=ld48ci1w
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
--- (10 headers 0 lines)---
-- SIP/7297-d075 is ringing
IPBX-TEST*CLI
-- SIP read from 10.101.51.248:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport=5060
From: DSOFT sip:[EMAIL PROTECTED];tag=as417ffda1
To: sip:[EMAIL PROTECTED]:2051;line=ld48ci1w;tag=okikki5eaw
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Contact: sip:[EMAIL PROTECTED]:2051;line=ld48ci1w
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
--- (10 headers 0 lines)---
-- SIP/7297-d075 is ringing
-- Channel 0/19, span 1 got hangup request
== Spawn extension (sip, 1745, 3) exited non-zero on 'Zap/19-1'
-- Hungup 'Zap/19-1'
IPBX-TEST*CLI
-- SIP read from 10.101.51.248:2051:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.101.51.252:5060;branch=z9hG4bK6587974f;rport=5060
From: DSOFT sip:[EMAIL PROTECTED];tag=as417ffda1
To: sip:[EMAIL PROTECTED]:2051;line=ld48ci1w;tag=okikki5eaw
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Contact: sip:[EMAIL PROTECTED]:2051;line=ld48ci1w
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
--- (10 headers 0 lines)---
-- SIP/7297-d075 is ringing
-- SIP/7303-336e answered Zap/6-1
-- Executing AGI(SIP/7303-336e, inscription_decroche.php) in new
stack
--
Re: [Asterisk-Users] ATA's ???
Hi Phil, if you want to use ATAs take a look at grandstream site...they are better than digium but you could use a card, TDM400 is excellent for analog lines and devices. Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, I'm currently in the process of building Asterisk for our new office and have hit a snag. We need two internal Analog lines for a modem and fax machine. Am I right in thinking I can use two ATA's, one on each piece of equipment which will then talk to Asterisk and route via our ISDN30? If the above is corrent could you recommend a good model? Thanks in advance. Phil. ___ --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] wcfxo md3200 problem...
hello all, i have a * 1.2.1, in a lab, only for test, with 4fxo clone - md3200 - intel537, connect to pstn. All work well, but, 1 once day 2 of this cards, stop make call, and receiv call thought. i kill the asterisk, remove modules, wcfxo and zaptel, mount the modules again, and start the *, for resolv the problem. No message in logs, the asterisk or system, dmesg, i use slackware 10.1 with kernel 2.4.29 any idea? thanks all,and sorry for bad english Alex, begin:vcard fn:Alex Montoanelli n:Montoanelli;Alex org;quoted-printable:Unetvale Internet =C2=B7 Agente Autorizado Brasil Telecom;Programador e Administrador de Redes email;internet:[EMAIL PROTECTED] tel;quoted-printable;work:48=C2=B73263 0013 tel;quoted-printable;cell:47=C2=B791498260 x-mozilla-html:TRUE version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38
I didn't find that exact message in the RFC's, but I did find something similar in RFC 3407 (http://www.rfc-archive.org/getrfc.php?rfc=3407), a=cdsc: 4 image udptl t38 Which means that the sender is capable of sending T.38 fax over UDP. I wouldn't worry about it unless you were trying to receive a T.38 fax over UDP, or it causes some other problem. If you need to get further into this, run sip debug from the console so you can see the entire SIP message in which this line appears. Giorgio Incantalupo wrote: Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo ___ --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] Monitoring
Hi asterisk and ser users, Is there a solution to monitor asterisk and ser with snmp ? Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spa3k and ISDN
What caller id method is used in spain? Is it before or after the ring. If you can set the ISDN termination box for UK caller id then the ID is sent before the first ring. on the sipura thats ETSI FSK with PR(UK) Chris - Original Message - From: Manuel Dominguez [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 27, 2006 11:29 AM Subject: [Asterisk-Users] Spa3k and ISDN Hello all, I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal analogue lines. The same number is assigned to these lines. These lines are connected to 2 spa3k registered to my asterisk box. When calls arrive, TR1 try to pass call to the first spa. If spa not takes the call immediately then try to pass to the other spa. The only configuration I found works is to put the parameter 'PSTN Answer Delay' to 0 in each spa. The problem is Call CID. I suppose the problem is that Asterisk not sees the CID because the spa takes several seconds to know. In the Spa status page appears the CID but never in the asterisk box or extensions. If somebody can help me it would be appreciate, Regards, Manuel ___ --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] Caller Presentation
Could someone please outline the differences between: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable Thank you -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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] Shared Line Appearance
Sean Cook wrote: Ok... I am having a serious brain fart this evening. IIRC, the next sip draft addresses shared lines and I thought I remembered something on the list about support for it in the near future. 'the next sip draft'? There are probably 150+ IETF drafts circulating regarding SIP functionality... is there a particular one you are referring to? I'm not aware of any effort to update/replace RFC3261, and shared line appearance wouldn't belong there anyway. Is there an implementation for shared line support in asterisk? I know that with hint I can watch line status... I just want to be able to pick up on an extension when ringing or jumping in on a call by punching the line. You are confusing shared line appearance with shared extension appearance. It is possible today to watch an extension's status and use the key on the phone to either call that extension (if it is not in use) or pick up a ringing call at that extension. With some creative dialplan programming it may also be possible to force any call that extension is involved in into a MeetMe and then join it... thereby joining the call. This is all 'shared extension appearance' stuff. Shared Line Appearance is much more complex to implement, but we are very seriously considering doing it in the near future, since there is so much demand. Keep in mind that you will _never_ be able to fully simulate a key system using Asterisk unless you seriously dumb down the Asterisk features that don't make sense for a key system... but we can at least get this part functional. Stay tuned :-) ___ --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] Announcement: Snom 360 with integrated XML O bjects
Hi.
Use massdeployment for putting the licenses on to your phones.
There is a setting called license_url you can use like the firmware update
URL, the macro {mac} will be replaced by the MAC address of the phone. So if
you provide the setting like this:
license_url: http://yourwebserver/{mac}.txt
all phones will download and install their licenses from this directory
automatically.
Ok, you need to send us a list of all MAC addresses and we will send you the
needed licenses.
BTW which firmware is already on the phones ? If it is 4.something and they
are working, the license is already on the phones. With firmware 4.0 you need
to have a license on the phone.
Best regards,
Sven
On Friday 27 January 2006 06:01, Colin Anderson wrote:
Is there any plans for a site license or some way to deploy the license a
little more elegantly? I have a lot of 360's!
I'm excited about this feature - it enables me to deploy some solutions
that I have been promising to my endusers. The two I have in mind are
Outlook calendar push to the display, and Outlook contact pull to the
directory. Some other ones will involve caller ID lookup to our CRM. If I
make progress along these lines, I will post results to the list.
-Original Message-
From: Christian Stredicke [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 26, 2006 9:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Announcement: Snom 360 with integrated XML
Objects
As far as the licence is concerned that is something that we introduced
in the 4.0 image and this is not against our customers (which would be
stupid). It shall protect us from clones.
The jump to the 5.0 is not about this licensing stuff, we just changed
the ramdisk and freed up more memory. I know this is not very pleasant,
and we cross fingers that this is the last time we have to do something
like this. But running out of memory is also not very pleasant!
Especially when new cool features ask for more memory!
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Colin Anderson
Sent: Thursday, January 26, 2006 2:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Announcement: Snom 360 with
integrated XML Objects
that is *very* cool. However, I am somewhat concerned about
being forced to license the firmware (even if it is free) can
you comment for the list the rationale behind forcing a
license and how this might affect Snom users who, say, want
to DOWN grade their firmware?
ps is there a timetable for supported, formally released
firmware version?
-Original Message-
From: Hirosh Dabui [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 26, 2006 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Announcement: Snom 360 with
integrated XML Objects
-BEGIN PGP SIGNED MESSAGE-
Hash: RIPEMD160
Dear user,
the new snom 360 is able to use services from standard web servers.
Users can deploy customized client services with snom 360 and
interact with other users via the keypad. The snom 360 will
use HTTP protocol from standard web servers, like Apache.
Typical services are:
~ 1. To-do lists
~ 2. Stock Information
~ 3. Weather
~ 4. Provisioning
~ 5. Agenda
~ 6. Telephone directory
For further information go to
http://snom.com/wiki/index.php/Xmlobjects
Note: *That is a pre-release, probably the software is still unstable*
Best regards,
Hirosh Dabui
- --
snom technology AG
Dipl.-Ing. Hirosh Dabui
PGP Key-ID: 0x30A34758
mailto:[EMAIL PROTECTED]
http://snom.com
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Version: GnuPG v1.4.2 (GNU/Linux)
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FCXMUdN9loiwy948EO8th9U=
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Re: [Asterisk-Users] TDM400 pinout
Hi I'm looking for a pinout for the above. Note this has what i'd call RJ45 sockets (or someone smart can correct me). I need to plug into BT (rj13?). Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12 sockets. I assume with the mention of BT, you're in the UK. The line is on pins 2+5 of the BT connector, which'd usually translate to the 2 inner pins of an RJ11 connector (pins 2+3). You should find an old modem cable will do the job fine. If your TDM400 really does have RJ45 sockets, then you'd expect the line to be on the middle pins (pins 4+5), similar to a modtap used in structured cabling environments. Regards, Chris Thanks, yes they are rj45, we have had rj12 in he past I look at the above. Like I said though, pity Digium dont supply the information on there site or with the cards, its a bit like everything in life today. We are only the customer, but we're expected to do the running around. Earlier versions of the TDM400 I believe were RJ45. They were changed to RJ11 I think I had heard at one point for compliance with some telco standards outside the US. But, in either case, yes, the middle pair is the active pair for your FXO/FXS ports on these cards whether RJ11 or RJ45. If I recall a conversation with Kevin correctly, all TDM400 cards prior to Rev I were rj11's. With the Rev J card, the connector was changed to something that looks very close to an rj45 but has a different number. The rj45-like connector was used for telco standard's compliance in one/more of the non-US countries (forgot which ones). At the time of the rj45-like change, there were also some minor pstn line filtering components added to address those same compliance issues. ___ --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] WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38
it means that your sender is capable of sending t38, but asterisk (without at a minimum the t38-patches for passthrough) is not capable of handling this. if you have reinvite for this channel allowed and your sender can send the fax over g711 asterisk will send a reinvite and the fax has a chance of getting through. afaik the warning is not an issue if you and your sender can reinvite to g711. greetings from graz, Am Freitag, 27. Januar 2006 11:48 schrieb Giorgio Incantalupo: Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo ___ --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 pgpYjWdUVtEjP.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA's ???
[EMAIL PROTECTED] wrote: Hi, I'm currently in the process of building Asterisk for our new office and have hit a snag. We need two internal Analog lines for a modem and fax machine. Am I right in thinking I can use two ATA's, one on each piece of equipment which will then talk to Asterisk and route via our ISDN30? If the above is corrent could you recommend a good model? Yes, that should work fine - I have a fax machine here connected to a Grandstream Handytone ATA-286 which (with recent firmware) performs faultlessly using G.711 aLaw. I'm not sure how well it will support higer speed modems, though... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] ATA's ???
Phil I have very good experience with the vegasteam ATAs devices.(you might also want to look @ sipura ATAs, since vegastream is doing an oem on there boxes) They support modem until v.90 speeds and faxes for g3. They are expensive, and again, work great and configure very easy joash From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, January 27, 2006 12:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ATA's ??? Hi, I'm currently in the process of building Asterisk for our new office and have hit a snag. We need two internal Analog lines for a modem and fax machine. Am I right in thinking I can use two ATA's, one on each piece of equipment which will then talk to Asterisk and route via our ISDN30? If the above is corrent could you recommend a good model? Thanks in advance. Phil. ___ --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] transfer, recording ...
Ronald Wiplinger wrote: does still not do the trick! Show your Dial command from extensions.conf file. -- Best regards, Bartosz Piec ___ --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] Best FXO hardware for home use
There is no doubt that given a particular scenario, anything won't work properly. This is not necessarily a problem with the SPA3000 or the TDM cards, this is much more of a phone line issue. Granted, those devices don't handle line issues as well as some other devices (such as the long loop issue you mentioned) but to write them off as being poor products I felt was a bit overkill. I have several very successful installs with TDM cards and SPA3000s and on the other hand I have an install that nothing seems to want to work with the PSTN lines that are there. So while I do agree with you on what the actual issue is, I don't think it is 100% fair to write off the SPA3000 in all cases. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com I have had nothing but problems with echo on my spa3000. I finally got it to work most of the time, but only after going back to the 3.1.3 firmware. The 3.1.5 and 3.1.7 firmware cause echo for me. And that happens to be one of the recommendations from the Sipura/Linksys support folks as well. Seems they know there is a problem, but aren't addressing it for some reason. ___ --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] Outgoing FXO and CDR
Henry Margies wrote: Hi all, When I do outgoing calls via my FXO card (TDM400, analog line), they get always marked ANSWERED in my CDR. I guess it is not that easy for fxo to determine if there is actually a call or just ringing. But anyway, is there a way to get this working right? If you are the USA, you can try to use callprogress=yes in zapata.conf, but the warnings above the entry still stand. -- 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] Packeting multiple GSM frames in one IP packet - Help needed.
[EMAIL PROTECTED] wrote: Hi, We have a task to reduce voice call bandwidth. IP+UDP+RTP are using 40 bytes per packet and for voice GSM FR 33 bytes. We are trying to reduce this bandwidth accommodating multiple GSM frames in one packet. If we want to use per packet 10 GSM frames how to do this using asterisk? Assume the sip client is able to split these packets in to individual GSM frames. Why don't you just buy some g729 licences? Is this a trunked IAX connection? -- 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] T38 providers
Have any providers started to offer T.38 yet? I am anxious to find a solution for faxing. -- Chris Mason -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Max concurrent calls
Whatever happened to Google? why don't people use that? Tha actual limit according to Google/wiki is/was 255 for zap channels: http://voip-info.org/tiki-index.php?page=Asterisk+dimensioning However, in that same post someone corrected it that it is no longer limited. On 1/27/06, Andrew Nowrot [EMAIL PROTECTED] wrote: Hi In my environment I have to connect 6 * boxes with each other so IAX is probably the best solution Thanks Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Help with sip setup because can't receive calls
Hi I basically allow=all and NAT=no for all the phones. but still can't see why I can't receive calls (i.e. in-bound)but I can make outbound calls. also there is no debuging on pbx for sip (unless it's outbound call). do you have anymore advice?thanks AmaMd Sani Johari [EMAIL PROTECTED] wrote: hi abc def, what type of voice codec that phone use. Maybe it can't support. I also have same problem my sip phone, when i change the voice codec from g729tog71
1 ulaw,
then it work find.also make sure wether your sip is behind the router or not.. nat=never or nat=1- Original Message - From: abc def To: asterisk-users@lists.digium.com Sent: Wednesday, January 25, 2006 8:58 PM Subject: [Asterisk-Users] Help with sip setup because can't receive calls Hi all, I readmany posts on asterisk mail site and been trying many different thingsbut still I can't get my sip phones to work with asterisk. I have a full blown-up voip netwok with two asterisk servers connected to pstn networkwith iax phones and cisco sccp phones which all work fine. however, I have been struggeling to configure my sip phones (polycom 601, Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip phones to anywhere else but not receive phone calls. I can see the phones on "sip show regi
stry"
and "sip show peers" but no track phone calls for sip. can you please shed some light on me how to go about solving this problem? thank you and best regards, Ama Do you Yahoo!?With a free 1 GB, there's more in store with Yahoo! Mail. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersNo virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.375 / Virus Database: 267.14.22/238 - Release Date:
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sk-Users
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Re: [Asterisk-Users] transfer, recording ...
Ronald Wiplinger wrote:
exten = 600,1,Dial(${PHONE_LOCAL},60,tr)
Type this:
exten = 600,1,Dial(${PHONE_LOCAL},60,tTwWr)
dial at 600 and see if this helps. If so, change all commands in that
way (tT is for transfer, wW is for recording).
You must also have sox installed for calls recording.
--
Best regards,
Bartosz Piec
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Re: [Asterisk-Users] Spa3k and ISDN
On Friday 27 January 2006 13:29, Manuel Dominguez wrote: Hello all, I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal analogue lines. The same number is assigned to these lines. These lines are connected to 2 spa3k registered to my asterisk box. When calls arrive, TR1 try to pass call to the first spa. If spa not takes the call immediately then try to pass to the other spa. The only configuration I found works is to put the parameter 'PSTN Answer Delay' to 0 in each spa. The problem is Call CID. I suppose the problem is that Asterisk not sees the CID because the spa takes several seconds to know. In the Spa status page appears the CID but never in the asterisk box or extensions. Connect a phone with a display through the FXS port and let the PSTN line to ring through to the VOIP. That way you can check do you receive callerid at all. You could play with the delay secs untill you are sure you see a callerid(probably only number). And most probably the callerid method should be ETSI. Any experience with as how a GSM CLIP is read by spa3000? Anyone? benchev ___ --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] Newbie SIP trunk question...
Hi Guys, We are using Grandstream BT-102 phones internally to talk directly to our SIP provider's SIP server. Each of our phones is configured with a CLI provided by our SIP provider. I have a couple of spare phones and about 5 spare CLI's, so I decided to set up AsteriskAtHome to see what it was like... I've got the phones working with Asterisk, but I'm unsure as to the best way to proceed to get Asterisk to negotiate SIP/CLI's, etc with out Provider's SIP server. i.e. Should I set up each of our CLI's as a SIP trunk or is there a better way to do things? Any help would be appreciated... Cheers, ojc -- ~~~ Owen Connolly Technical Director http://www.networkarchitects.ie ~~~ ___ --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] T38 providers
On 1/27/06, Chris Mason (Lists) [EMAIL PROTECTED] wrote: Have any providers started to offer T.38 yet? I am anxious to find a solution for faxing. commpartners does offer it. I haven't personally used it yet, but I know they offer the service. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] AAH out bound routing problem
Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700 -- Got SIP response 488 Not acceptable here back from(PeerIP) -- SIP/easycall-838e is circuit-busy ram ___ --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] Help with sip setup because can't receive calls
there is no error message coming up on the pbx for in-bound calls (there is only debugging messages for outbound calls).thanks in advance for any hint or suggestion. AmaI just post my configuration file here for sip phone: extensions.conf-[globals] [default]include = incominginclude = outgoinginclude = iaxinculde = sipinclude = sccp[sip]exten = 2171,1,Dial(SIP/stargate1,20);exten = 2171,1,Dial(SIP/2171,20)exten = 2171,2,Hangupexten = 2172,1,Dial(SIP/stargate2,20);exten = 2172,1,Dial(SIP/2172,20)exten = 2172,2,Hangupexten = 2173,1,Dial(SIP/stargate3,20);exten = 2173,1,Dial(SIP/2173,20)exten = 2173,2,Hangup [sccp] [skinny] [incoming]exten =
;
_214943[5-9]6,1,Dial(SIP/stargate3)exten = _214943[5-9]6,2,Hangup [outgoing]exten = _,1,Dial(Zap/g1/${EXTEN})exten = _,2,Hangup- sip.conf-[general]context=default ; Default context for incoming calls ; Set this to your host name or domain namebindport=5060 ; UDP Port to bind to (SIP standard port is
5060)bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup=yes ; Enable DNS SRV lookups on outbound calls register = stargate1:[EMAIL PROTECTED]/2171register = stargate2:[EMAIL PROTECTED]/2172register = stargate3:[EMAIL PROTECTED]/2173;-- NAT SUPPORT nat=no ; Global NAT settings (Affects all peers and
users) [local_sip]type=friendhost=10.47.200.136context=default [stargate1] ;cisco 9760;[2171]type=friendhost=dynamic ;10.47.200.140 ;dynamicdefaultip=10.47.200.140username=stargate1secret=xxxcallerid="21495071" 2171allow=allqualify=200nat=nodefaultip=10.47.200.140 [stargate2] ;Polycom 601;[2172]type=friendhost=dynamic ;10.47.200.141 ;dynamicdefaultip=10.47.200.141username=xxxsecret=2stargatecallerid="21495072" 2172allow=allqualify=200nat=nodefaultip=10.47.200.141 [stargate3] ;Aastra 480i;[2173]type=friendhost=dynamic ;10.47.200.137 ;dynamicdefaultip=10.47.200.137username=stargate3callerid="starg
ate3"
2173secret=xxxallow=allqualify=200nat=nodefaultip=10.47.200.137 [EMAIL PROTECTED] wrote: What error do you get when trying to call the SIP phones?PaulH - Original Message - From: abc def
To: asterisk-users@lists.digium.com Sent: Wednesday, January 25, 2006 11:58 PM Subject: [Asterisk-Users] Help with sip setup because can't receive calls Hi all, I readmany posts on asterisk mail site and been trying many different thingsbut still I can't get my sip phones to work with asterisk. I have a full blown-up voip netwok with two asterisk servers connected to pstn networkwith iax phones and cisco sccp phones which all work fine. however, I have been struggeling to configure my sip phones (polycom 601, Aastra 480i and cisco 9760) to work with asterisk. I can call out from sip phones to anywhere else but not receive phone calls. I can see
the
phones on "sip show registry" and "sip show peers" but no track phone calls for sip. can you please shed some light on me how to go about solving this problem? thank you and best regards, Ama Do you Yahoo!?With a free 1 GB, there's more in store with Yahoo! Mail. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options
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Re: [Asterisk-Users] transfer, recording ...
Bartosz Piec wrote:
Ronald Wiplinger wrote:
exten = 600,1,Dial(${PHONE_LOCAL},60,tr)
Type this:
exten = 600,1,Dial(${PHONE_LOCAL},60,tTwWr)
dial at 600 and see if this helps. If so, change all commands in that
way (tT is for transfer, wW is for recording).
You must also have sox installed for calls recording.
No, it did not change anything!
cannot transfer!
bye
Ronald Wiplinger
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[Asterisk-Users] Re: ztdummy
It does use the same kernel for everything. It's a specially modified kernel for the VPS support. I guess the only way to see if ztdummy works in the VPS is to try it. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 27, 2006 2:54 AM Subject: Asterisk-Users Digest, Vol 18, Issue 175 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Chan_capi on builds 79558320 strangeness ([EMAIL PROTECTED]) 2. S100-FX v2.0 (Mike Hammett) 3. Alex Tew interview made possible because of Simon @ Simwood eSMS (Ronald Lewis) 4. How to put peers into Realtime (Ronald Wiplinger) 5. Re: transfer, recording ... (Ronald Wiplinger) 6. paging agi (Jeremy) 7. Re: Chan_capi on builds 79558320 strangeness (Armin Schindler) 8. Max concurrent calls (Andrew Nowrot) 9. Re: ztdummy (Tzafrir Cohen) 10. Re: Max concurrent calls (Zoa) 11. Asterisk authorization (Sam Tam) 12. RE: Dynamically disabling echo cancellation (Zap). (Koopmann, Jan-Peter) 13. Re: Max concurrent calls (Jean-Michel Hiver) 14. Re: paging agi (Tzafrir Cohen) 15. RE: Chan_capi on builds 79558320 strangeness ([EMAIL PROTECTED]) 16. ASTPP (Ronald Ramos) 17. Re: Max concurrent calls (Andrew Nowrot) 18. Offtopic: Auto provioning Snom 360 (Erik) 19. RE: Announcement: Snom 360 with integrated XML Objects (Dovid Bender) 20. Re: Asterisk authorization (Umair Bari) 21. Re: Max concurrent calls (Andrew Nowrot) 22. Packeting multiple GSM frames in one IP packet - Help needed. ([EMAIL PROTECTED]) 23. Re: Max concurrent calls (Jean-Michel Hiver) 24. Re: Max concurrent calls (Jean-Michel Hiver) 25. DeadAGI and Hangup on channel (Grigoriy Puzankin) -- Message: 9 Date: Fri, 27 Jan 2006 09:58:43 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ztdummy To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Thu, Jan 26, 2006 at 03:10:09PM -0600, Mike Hammett wrote: I'm running a VPS and I need to pass the device drivers from the host OS to the VPS. What files do I need to pass through for ztdummy to work? I'm assuming they're in /dev/zap, but I'm not sure which ones are needed. ztdummy (of kernel 2.6) should not require anything from the host. However are you sure you can use different kernels for the host and the guest with your VPS? It does generate a load of 1000 interrupts per second. This means tha tyou always need CPU time. And a timely response of it. -- 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] OT?: International number parsing
Can anyone shed some light on rules that might make the task of parsing the country code and city codes from a dialed number in the CDRs? I know that there is almost never a case where a concatenated country and city code could overlap with another country code, but what about city codes and local numbers? Is it possible for a concatenated city code and local number to match another city code in the same country? I already have the table of country and city codes built. Are there holes in this theory; 1. Starting after the international dialing code, find the longest match for country code. 2. Starting after the country code from step 1, find the longest match for city code within that countries table of city codes. 3. The rest is the local number. Are there known exceptions? Am I reinventing the wheel rather than finding the right already existing resource? ___ --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] External IAX2 phone defined as internal behaving as from PSTN
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls from 1055 get picked up as if it were an external call
(see below) and goes straight to the ring group macro.
The same phone either on the same internal network to the asterisk or on
a VPN to said network work fine. Obviously asterisk thinks this call is
external.
How do change this?
asterisk1*CLI
-- Accepting AUTHENTICATED call from 99.99.99.212: {faked}
requested format = g729,
requested prefs = (),
actual format = g729,
host prefs = (g729|gsm|ulaw|alaw),
priority = mine
-- Executing Macro(IAX2/1055-4,
rg-group|ringall|60||1000-1001-1007-1050-1450-1600) in new stack
-- Executing Macro(IAX2/1055-4, user-callerid) in new stack
-- Executing DBget(IAX2/1055-4, AMPUSER=DEVICE/1055/user) in new
stack
-- DBget: varname=AMPUSER, family=DEVICE, key=1055/user
-- DBget: set variable AMPUSER to 1055
-- Executing DBget(IAX2/1055-4,
AMPUSERCIDNAME=AMPUSER/1055/cidname) in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=1055/cidname
-- DBget: set variable AMPUSERCIDNAME to EXTERNAL-CALLER
-- Executing GotoIf(IAX2/1055-4, 0?5) in new stack
-- Executing SetCallerID(IAX2/1055-4, EXTERNAL-CALLER 1055)
in new stack
-- Executing NoOp(IAX2/1055-4, Using CallerID EXTERNAL-CALLER
1055) in new stack
-- Executing GotoIf(IAX2/1055-4, 0?4:3) in new stack
-- Goto (macro-rg-group,s,3)
-- Executing SetCIDName(IAX2/1055-4, EXTERNAL-CALLER) in new
stack
-- Executing SetVar(IAX2/1055-4, RGPREFIX=) in new stack
-- Executing SetCIDName(IAX2/1055-4, EXTERNAL-CALLER) in new
stack
-- Executing SetVar(IAX2/1055-4, RecordMethod=Group) in new
stack
-- Executing Macro(IAX2/1055-4, record-enable|1|Group) in new
stack
-- Executing GotoIf(IAX2/1055-4, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(IAX2/1055-4,
recordingcheck|20060126-193156|1138303916.91) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(IAX2/1055-4, No recording needed) in new stack
-- Executing SetVar(IAX2/1055-4, RingGroupMethod=ringall) in new
stack
-- Executing Macro(IAX2/1055-4,
dial|60|tr|1000-1001-1007-1050-1450-1600) in new stack
-- Executing GotoIf(IAX2/1055-4, 1?4:2) in new stack
-- Goto (macro-dial,s,4)
-- Executing AGI(IAX2/1055-4, dialparties.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
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RE: [Asterisk-Users] Announcement: Snom 360 with integrated XML O bjects
Aha, I see it's 4.1, cool. So I just have to do a straight upgrade to 5.0
and I have this new toy to play with, correct?
-Original Message-
From: Sven Fischer (support) [mailto:[EMAIL PROTECTED]
Sent: Friday, January 27, 2006 5:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Announcement: Snom 360 with integrated XML
O bjects
Hi.
Use massdeployment for putting the licenses on to your phones.
There is a setting called license_url you can use like the firmware update
URL, the macro {mac} will be replaced by the MAC address of the phone. So if
you provide the setting like this:
license_url: http://yourwebserver/{mac}.txt
all phones will download and install their licenses from this directory
automatically.
Ok, you need to send us a list of all MAC addresses and we will send you the
needed licenses.
BTW which firmware is already on the phones ? If it is 4.something and they
are working, the license is already on the phones. With firmware 4.0 you
need
to have a license on the phone.
Best regards,
Sven
On Friday 27 January 2006 06:01, Colin Anderson wrote:
Is there any plans for a site license or some way to deploy the license a
little more elegantly? I have a lot of 360's!
I'm excited about this feature - it enables me to deploy some solutions
that I have been promising to my endusers. The two I have in mind are
Outlook calendar push to the display, and Outlook contact pull to the
directory. Some other ones will involve caller ID lookup to our CRM. If I
make progress along these lines, I will post results to the list.
-Original Message-
From: Christian Stredicke [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 26, 2006 9:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Announcement: Snom 360 with integrated XML
Objects
As far as the licence is concerned that is something that we introduced
in the 4.0 image and this is not against our customers (which would be
stupid). It shall protect us from clones.
The jump to the 5.0 is not about this licensing stuff, we just changed
the ramdisk and freed up more memory. I know this is not very pleasant,
and we cross fingers that this is the last time we have to do something
like this. But running out of memory is also not very pleasant!
Especially when new cool features ask for more memory!
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Colin Anderson
Sent: Thursday, January 26, 2006 2:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Announcement: Snom 360 with
integrated XML Objects
that is *very* cool. However, I am somewhat concerned about
being forced to license the firmware (even if it is free) can
you comment for the list the rationale behind forcing a
license and how this might affect Snom users who, say, want
to DOWN grade their firmware?
ps is there a timetable for supported, formally released
firmware version?
-Original Message-
From: Hirosh Dabui [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 26, 2006 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Announcement: Snom 360 with
integrated XML Objects
-BEGIN PGP SIGNED MESSAGE-
Hash: RIPEMD160
Dear user,
the new snom 360 is able to use services from standard web servers.
Users can deploy customized client services with snom 360 and
interact with other users via the keypad. The snom 360 will
use HTTP protocol from standard web servers, like Apache.
Typical services are:
~ 1. To-do lists
~ 2. Stock Information
~ 3. Weather
~ 4. Provisioning
~ 5. Agenda
~ 6. Telephone directory
For further information go to
http://snom.com/wiki/index.php/Xmlobjects
Note: *That is a pre-release, probably the software is still unstable*
Best regards,
Hirosh Dabui
- --
snom technology AG
Dipl.-Ing. Hirosh Dabui
PGP Key-ID: 0x30A34758
mailto:[EMAIL PROTECTED]
http://snom.com
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Version: GnuPG v1.4.2 (GNU/Linux)
iD8DBQFD2Q6YAO47/DCjR1gRA6REAJ4iSyot8OhFVDt0/C2I7KFoRCP18ACeNGau
FCXMUdN9loiwy948EO8th9U=
=Qntp
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Re: [Asterisk-Users] AAH out bound routing problem
On Fri, January 27, 2006 15:13, ram said: Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700 -- Got SIP response 488 Not acceptable here back from (PeerIP) -- SIP/easycall-838e is circuit-busy ram Most likely the telno provided (19197543700) is not compatible with what they expect... Maybe you need to att digits (Perhaps 0019197543700) or remove digits? Or maybe you're not authenticated ? We'll need more info to be able to assist any further... To begin with it would help to know what configuration they expect... -- 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] External IAX2 phone defined as internal behaving as from PSTN
On Fri, January 27, 2006 16:09, Ian Cowley said: Have [EMAIL PROTECTED] 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring group macro. The same phone either on the same internal network to the asterisk or on a VPN to said network work fine. Obviously asterisk thinks this call is external. How do change this? SNIP The actual iax.conf part pertaining to this phone might be helpful here... -- 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] Good provider of Polycom Phones (mostly for access to latest/greatest firmware)
Hi, I've ordered a few IP501s from PC Connection, basically since we have an account with them. I like the phones for what they do, and now would like establish a relationship with a reseller that can give us maintenance and access to the most current firmware. What are some good resellers out there? Regards, --- Gavin Adams VP of Technology Promisant (USA) Inc. Email: [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] AAH out bound routing problem
Ram, On my AAH the stock dial plan requires a 9 first. For kicks, try dialing 919197543700 and see what you get. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ram Sent: Friday, January 27, 2006 6:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] AAH out bound routing problem Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700 -- Got SIP response 488 Not acceptable here back from(PeerIP) -- SIP/easycall-838e is circuit-busy ram ___ --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] using sangoma cards as a timesource?
We use APIC on all servers, so interrupt sharing is not an issue :) On Jan 26, 2006, at 3:02 PM, Damon Estep wrote: And in some (many) cases it will do so while sharing an interrupt with a NIC and disk controller! We run sangoma a104 cards in Dell SC1425 1U servers with great success under heavy load. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Thursday, January 26, 2006 5:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] using sangoma cards as a timesource? Short answer: Yes Long answer: They use the zaptel drivers and are recognized as a Zaptel device. You do have to load and configure the Sangoma wanpipe drivers first, but in the end it'll function as a timing source just like a Digium card MATT--- On 1/26/06, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi building a new setup, we want to try using sangoma cards. can these be used as time sources the same way as TE410Ps? thanks roy ___ --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 ___ --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] SER redirect
hello, can someone help me with ser redirect to asterisk. any help appreciated. Thanks, AA ___ --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] External IAX2 phone defined as internal behaving as from PSTN
-Original Message-
From: Ian Cowley
Sent: 27 January 2006 15:10
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] External IAX2 phone defined as internal
behaving as from PSTN
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls from 1055 get picked up as if it were an external call
(see below) and goes straight to the ring group macro.
The same phone either on the same internal network to the asterisk or on
a VPN to said network work fine. Obviously asterisk thinks this call is
external.
How do change this?
asterisk1*CLI
-- Accepting AUTHENTICATED call from 99.99.99.212: {faked}
requested format = g729,
requested prefs = (),
actual format = g729,
host prefs = (g729|gsm|ulaw|alaw),
priority = mine
-- Executing Macro(IAX2/1055-4,
rg-group|ringall|60||1000-1001-1007-1050-1450-1600) in new stack
-- Executing Macro(IAX2/1055-4, user-callerid) in new stack
-- Executing DBget(IAX2/1055-4, AMPUSER=DEVICE/1055/user) in new
stack
-- DBget: varname=AMPUSER, family=DEVICE, key=1055/user
-- DBget: set variable AMPUSER to 1055
-- Executing DBget(IAX2/1055-4,
AMPUSERCIDNAME=AMPUSER/1055/cidname) in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=1055/cidname
-- DBget: set variable AMPUSERCIDNAME to EXTERNAL-CALLER
-- Executing GotoIf(IAX2/1055-4, 0?5) in new stack
-- Executing SetCallerID(IAX2/1055-4, EXTERNAL-CALLER 1055)
in new stack
-- Executing NoOp(IAX2/1055-4, Using CallerID EXTERNAL-CALLER
1055) in new stack
-- Executing GotoIf(IAX2/1055-4, 0?4:3) in new stack
-- Goto (macro-rg-group,s,3)
-- Executing SetCIDName(IAX2/1055-4, EXTERNAL-CALLER) in new
stack
-- Executing SetVar(IAX2/1055-4, RGPREFIX=) in new stack
-- Executing SetCIDName(IAX2/1055-4, EXTERNAL-CALLER) in new
stack
-- Executing SetVar(IAX2/1055-4, RecordMethod=Group) in new
stack
-- Executing Macro(IAX2/1055-4, record-enable|1|Group) in new
stack
-- Executing GotoIf(IAX2/1055-4, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(IAX2/1055-4,
recordingcheck|20060126-193156|1138303916.91) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(IAX2/1055-4, No recording needed) in new stack
-- Executing SetVar(IAX2/1055-4, RingGroupMethod=ringall) in new
stack
-- Executing Macro(IAX2/1055-4,
dial|60|tr|1000-1001-1007-1050-1450-1600) in new stack
-- Executing GotoIf(IAX2/1055-4, 1?4:2) in new stack
-- Goto (macro-dial,s,4)
-- Executing AGI(IAX2/1055-4, dialparties.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
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Please reply to this message to let us know you received it in error and also
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This disclaimer confirms that MessageLabs have swept e-mail and attachments for
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[Asterisk-Users] Problems with MFC/R2 in Brazil
I have installed a Digium card TE210P and unicall for use MFC/R2. I think
that it´s all right but I can´t make and receive calls. I´m using asterisk 2.1
with the patch made by José P. Leitão andthe follow libs:
libsupertone-0.0.2libunicall-0.0.3libmfcr2-0.0.3zaptel
2.1
My number is 34318300. The Telco send me only 8300. I see that
I receive from the Telco the first digit (8) and my asterisk answer 5, but the
Telco doesn´t receive my digit.My configs are below:
Itried to change the timer in mfcr2.c to 2. I tried a lot of
combinations in protocolvariant but, no sucess.
Please help me. Thanks a
lot.*unicall.conf*;call telephony
channel driver; Sample configuration file
[channels]loglevel=255language=brcontext=defaultusecallerid=yeshidecallerid=norestrictcid=nocallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=nocallreturn=noechocancel=yesechocancelwhenbridged=yes
echotraining=yesechotraining=800relaxdtmf=yesrxgain=0.0txgain=
0.0callgroup=1pickupgroup=1immediate=nocallerid=asreceivedamaflags=defaultaccountcode=line-E1faxdetect=nomusiconhold=defaultprotocolclass=mfcr2protocolvariant=br,10,4protocolend=cpegroup=1channel
= 1-15channel = 17-31
*zaptel.conf*span=1,1,0,cas,hdb3cas=1-15:1101cas=17-31:1101loadzone
= brdefaultzone=br
;extensions.conf*[general]static=yeswriteprotect=no
[default]exten = _,1,SetCallerID("Betha
Sistemas",4834318300)exten =
_,2,Dial(Unicall/g1/${EXTEN},60,t)exten =
_3XXX,1,Macro(sipiax,IAX2/${EXTEN})
exten=8300,1,Goto(telefonista,s,1) ;ligação cai na fila
datelefonistaexten=8301,1,Macro(sipiax,IAX2/3001) ;ligação
cai diretamente noramal
desejadoexten=8302,1,Goto(suporte_tributos,s,1) ;ligação cai na
fila dosuporte tributosexten=8303,1,Goto(telefonista,s,1)
;ligação cai na fila
datelefonistaexten=8304,1,Goto(telefonista,s,1) ;ligação cai
na fila datelefonista
;Fila de Atendimento
Telefonista[telefonista]exten=s,1,Answer(2)exten=s,2,SetMusicOnHold(default)exten=s,3,Queue(telefonista)
;Fila de Atendimento Suporte
Tributos[suporte_tributos]exten=s,1,Answer()exten=s,2,SetMusicOnHold(default)exten=s,3,DigitTimeout,5exten=s,4,ResponseTimeout,10exten=s,5,SetVar(CALLFILENAME=i${CALLERIDNUM}-${TIMESTAMP});exten=s,5,Background(fila_de_atendimento)exten=s,6,Queue(suporte-tributos)
;Login para a fila de
atendimentoexten=801,1,Wait,1exten=801,2,AgentLogin()[macro-sipiax]exten=s,1,SetLanguage(${LANG})exten=s,2,SetCallerId(${CALLERID})exten=s,3,Dial(${ARG1},20,Ttr)exten=s,4,Goto(s-${DIALSTATUS},1)exten=s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})exten=s-NOANSWER,2,Hangup()exten=s-CHANUNAVAIL,1,Voicemail(u${MACRO_EXTEN})
;O ramal
estáindisponívelexten=s-CHANUNAVAIL,2,Hangup()exten=s-BUSY,1,Voicemail(b${MACRO_EXTEN});o
ramal não
estáocupadodoexten=s-BUSY,2,Hangup()exten=s-CONGESTION,1,Voicemail(b${MACRO_EXTEN});o
ramal não está´disponível
exten=s-CONGESTION,2,Hangup()
Darlon Ferreira BortoliniRede/DesenvolvimentoBetha
SistemasFone (48) 431-0750/Ramal 1000
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Re: [Asterisk-Users] Shared Line Appearance
On 1/27/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: Sean Cook wrote: Is there an implementation for shared line support in asterisk? I know that with hint I can watch line status... I just want to be able to pick up on an extension when ringing or jumping in on a call by punching the line. You are confusing shared line appearance with shared extension appearance. It is possible today to watch an extension's status and use the key on the phone to either call that extension (if it is not in use) or pick up a ringing call at that extension. With some creative dialplan programming it may also be possible to force any call that extension is involved in into a MeetMe and then join it... thereby joining the call. This is all 'shared extension appearance' stuff. Pick up a ringing call at that extension? I can see how you would do the rest of the things you mentioned, but how would you pick up a ringing call going to that extension? Shared Line Appearance is much more complex to implement, but we are very seriously considering doing it in the near future, since there is so much demand. Keep in mind that you will _never_ be able to fully simulate a key system using Asterisk unless you seriously dumb down the Asterisk features that don't make sense for a key system... but we can at least get this part functional. Stay tuned :-) That would be great to see. ___ --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] External IAX2 phone defined as internal behaving as from PSTN
Iax.conf [general] ;bindport = 4569 ; Port to bind to (IAX is 4569) bindport = 5036 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=g729 ; 4 simultaneous allowed allow ilbc ; prefered for iax2 allow=gsm ; 13 Kbps (full rate), 20ms frame size allow=ulaw ;(g711)64 Kbps, sample-based allow=alaw ;(g711)64 Kbps, sample-based mailboxdetail=yes jitterbuffer=yes context=from-internal #include iax_additional.conf #include iax_custom.conf iax_additional.conf [1055] username=1055 type=friend secret=# record_out=Adhoc record_in=Adhoc qualify=yes port=4569 notransfer=yes [EMAIL PROTECTED] host=dynamic context=from-internal callerid=device 1055 Regards ianC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters (Asterisk) Sent: 27 January 2006 15:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] External IAX2 phone defined as internal behaving as from PSTN On Fri, January 27, 2006 16:09, Ian Cowley said: Have [EMAIL PROTECTED] 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring group macro. The same phone either on the same internal network to the asterisk or on a VPN to said network work fine. Obviously asterisk thinks this call is external. How do change this? SNIP The actual iax.conf part pertaining to this phone might be helpful here... -- 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 This Inbound email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com This email has been scanned for all viruses by the MessageLabs Email Security System. For more information on a proactive email security service working around the clock, around the globe, visit http://www.messagelabs.com ___ This e-mail message and any attachments may be confidential and may also be a privileged communication. It is intended solely for the person(s)to whom it is addressed. If you are not the intended addressee of the message you must take no action based on it. Please reply to this message to let us know you received it in error and also delete the message from your system. This disclaimer confirms that MessageLabs have swept e-mail and attachments for viruses on behalf of RSM Moffat Ltd. However it does not guarantee that either are virus-free and accepts no liability for any damage sustained as a result of viruses. RSM MOFFAT LTD Registered in UK. [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] No IN and OUT on ISDN line at the same time?
Hi, I like to forward an incoming call on an ISDN line to my mobile phone. Since ISDN offers two channels, I thought that this should work, but Asterisk tells me, that there is no channel available. There is no one else using this line, so guess I made a mistake in the configuration or it might not work for another reason. Here's the CLI output , the capi.conf and extensions.conf. 83086921 is the number that I have dialed and zzz is the number of my mobile phone. I even tried to setup up [ISDN2] and setting devices=1, too, but it didn't change anything. *CLI == ISDN1: Incoming call 'xx' - '83086921' -- Executing Dial(CAPI/ISDN1/83086921-0, CAPI/ISDN1/|20|tr) in new stack -- Called ISDN1/zz CAPI INFO 0x34a2: No circuit / channel available -- CAPI/ISDN1/zzz-1 is circuit-busy == ISDN1: CAPI Hangingup == Everyone is busy/congested at this time ;/etc/asterisk/capi.conf ; chan_capi_cm with EICON 2BRI [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de [ISDN1] incomingmsn=83086920,83086921 isdnmode=msn group=1 controller=1 softdtmf=1 context=demo echosquelch=1 echocancel=yes echotail=64 callgroup=1 devices=2 ;/etc/asterisk/extensions.conf [demo] exten = 83086921,1,Dial(CAPI/ISDN1/zzz,20,tr) exten = 83086921,2,hangup If I forward the call to a SIP phone, or start a call to my mobile phone from a SIP phone, the phones ring. Thanks for any help or hints, Ralf -- ___ Play 100s of games for FREE! http://games.mail.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] Digium Wildcard TDM400P call pickup timing
I have an analogue trunk to an ATT Definity.
It has a DISA context defined.
From a Definity handset call the analogue port extension 1008 and wait
for dial tone from asterisk. It takes between 34 rings.
Likewise from Asterisk SIP handset PBX Access NoPBX Extn takes
nearly 10 secs to ring.
Is this configurable?
Ian Cowley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian Cowley
Sent: 27 January 2006 15:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] External IAX2 phone defined as internal
behavingas from PSTN
-Original Message-
From: Ian Cowley
Sent: 27 January 2006 15:10
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] External IAX2 phone defined as internal
behaving as from PSTN
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls from 1055 get picked up as if it were an external call
(see below) and goes straight to the ring group macro.
The same phone either on the same internal network to the asterisk or on
a VPN to said network work fine. Obviously asterisk thinks this call is
external.
How do change this?
asterisk1*CLI
-- Accepting AUTHENTICATED call from 99.99.99.212: {faked}
requested format = g729,
requested prefs = (),
actual format = g729,
host prefs = (g729|gsm|ulaw|alaw),
priority = mine
-- Executing Macro(IAX2/1055-4,
rg-group|ringall|60||1000-1001-1007-1050-1450-1600) in new stack
-- Executing Macro(IAX2/1055-4, user-callerid) in new stack
-- Executing DBget(IAX2/1055-4, AMPUSER=DEVICE/1055/user) in new
stack
-- DBget: varname=AMPUSER, family=DEVICE, key=1055/user
-- DBget: set variable AMPUSER to 1055
-- Executing DBget(IAX2/1055-4,
AMPUSERCIDNAME=AMPUSER/1055/cidname) in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=1055/cidname
-- DBget: set variable AMPUSERCIDNAME to EXTERNAL-CALLER
-- Executing GotoIf(IAX2/1055-4, 0?5) in new stack
-- Executing SetCallerID(IAX2/1055-4, EXTERNAL-CALLER 1055)
in new stack
-- Executing NoOp(IAX2/1055-4, Using CallerID EXTERNAL-CALLER
1055) in new stack
-- Executing GotoIf(IAX2/1055-4, 0?4:3) in new stack
-- Goto (macro-rg-group,s,3)
-- Executing SetCIDName(IAX2/1055-4, EXTERNAL-CALLER) in new
stack
-- Executing SetVar(IAX2/1055-4, RGPREFIX=) in new stack
-- Executing SetCIDName(IAX2/1055-4, EXTERNAL-CALLER) in new
stack
-- Executing SetVar(IAX2/1055-4, RecordMethod=Group) in new
stack
-- Executing Macro(IAX2/1055-4, record-enable|1|Group) in new
stack
-- Executing GotoIf(IAX2/1055-4, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(IAX2/1055-4,
recordingcheck|20060126-193156|1138303916.91) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(IAX2/1055-4, No recording needed) in new stack
-- Executing SetVar(IAX2/1055-4, RingGroupMethod=ringall) in new
stack
-- Executing Macro(IAX2/1055-4,
dial|60|tr|1000-1001-1007-1050-1450-1600) in new stack
-- Executing GotoIf(IAX2/1055-4, 1?4:2) in new stack
-- Goto (macro-dial,s,4)
-- Executing AGI(IAX2/1055-4, dialparties.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
This email has been scanned for all viruses by the MessageLabs Email
Security System. For more information on a proactive email security
service working around the clock, around the globe, visit
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___
This e-mail message and any attachments may be confidential and may also
be a privileged communication.
It is intended solely for the person(s)to whom it is addressed. If you
are not the intended addressee of the message you must take no action
based on it.
Please reply to this message to let us know you received it in error and
also delete the message from your system.
This disclaimer confirms that MessageLabs have swept e-mail and
attachments for viruses on behalf of RSM Moffat Ltd.
However it does not guarantee that either are virus-free and accepts no
liability for any damage sustained as a result of viruses.
RSM MOFFAT LTD Registered in UK. [EMAIL PROTECTED]
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RE: [Asterisk-Users] Polycom 501 horrible echo
I've been running 1.6.4.0064 for the last few weeks.. I've had no problems with it, I haven't done a whole lot of speaker phone with it yet though.. Once my IP4000 reboots It'll be running it as well so that will be a good test. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Herring Sent: January 26, 2006 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom 501 horrible echo Now I'm really confused... 1.6.3 is on the Polycom Website as the latest... I'm running 1.6.2.0041 according to my phone. Which firmware worked for you? At 04:04 PM 1/26/2006, Ron Senykoff wrote: We also have noticed a poor server config can cause this in testing. Noticed when I had one person building * servers using Debian. Had them rebuilt with FC4 and have no issues - yet:) I recently upgraded all our phones to the latest Polycom firmware 1.6.2 and went from great speakerphone to tons of feedback. I would hate to have to go back to the old firmware. Although Polycom recommends keeping the older bootrom unless you need https provisioning, I'm going to try the new bootrom and see if it fixes the problem. This is being experienced across 3 corporate offices with 3 separate Asterisk servers. And I have to reiterate... all was good until the firmware upgrade. -Ron ___ --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
[Asterisk-Users] SIP incoming calls
To which context of the dial-plan does asterisk tries to match incoming calls when acting as a sip client? To be more specific: In extensions.conf Under which context should I place exten = 648064,1,Dial(TECH/peer) for an entry like this register = 648064:[EMAIL PROTECTED]/648064 ? This is because I want to match one sip client to one context, and another sip client into another context. Is it possible? What is the correct way to do it?? Thanks, Alejandro ___ --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] External IAX2 phone defined as internal behaving as from PSTN
On Fri, January 27, 2006 17:23, Ian Cowley said: Iax.conf [general] ;bindport = 4569 ; Port to bind to (IAX is 4569) bindport = 5036 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=g729 ; 4 simultaneous allowed allow ilbc ; prefered for iax2 allow=gsm ; 13 Kbps (full rate), 20ms frame size allow=ulaw ;(g711)64 Kbps, sample-based allow=alaw ;(g711)64 Kbps, sample-based mailboxdetail=yes jitterbuffer=yes context=from-internal #include iax_additional.conf #include iax_custom.conf iax_additional.conf [1055] username=1055 type=friend secret=# record_out=Adhoc record_in=Adhoc qualify=yes port=4569 notransfer=yes [EMAIL PROTECTED] host=dynamic context=from-internal callerid=device 1055 Regards ianC Looks like you are using AMP / [EMAIL PROTECTED] As far as I can tell, this should work correctly... There might be something going on in the translation by the Checkpoint NAT control... Have you tried iax2 debug to see what it is receiving? the first few packets should give you sufficient information... Good luck! -- 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] Polycom 501 horrible echo
I've been running 1.6.4.0064 for the last few weeks.. I've had no problems with it, I haven't done a whole lot of speaker phone with it yet though.. Once my IP4000 reboots It'll be running it as well so that will be a good test. Which bootrom version are you using? -Ron ___ --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] No IN and OUT on ISDN line at the same time?
The card is telling: CAPI INFO 0x34a2: No circuit / channel available so the other channel must be in use by something else. Maybe another device on the ISDN line? Armin On Sat, 28 Jan 2006, Ralf Mueller wrote: Hi, I like to forward an incoming call on an ISDN line to my mobile phone. Since ISDN offers two channels, I thought that this should work, but Asterisk tells me, that there is no channel available. There is no one else using this line, so guess I made a mistake in the configuration or it might not work for another reason. Here's the CLI output , the capi.conf and extensions.conf. 83086921 is the number that I have dialed and zzz is the number of my mobile phone. I even tried to setup up [ISDN2] and setting devices=1, too, but it didn't change anything. *CLI == ISDN1: Incoming call 'xx' - '83086921' -- Executing Dial(CAPI/ISDN1/83086921-0, CAPI/ISDN1/|20|tr) in new stack -- Called ISDN1/zz CAPI INFO 0x34a2: No circuit / channel available -- CAPI/ISDN1/zzz-1 is circuit-busy == ISDN1: CAPI Hangingup == Everyone is busy/congested at this time ;/etc/asterisk/capi.conf ; chan_capi_cm with EICON 2BRI [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de [ISDN1] incomingmsn=83086920,83086921 isdnmode=msn group=1 controller=1 softdtmf=1 context=demo echosquelch=1 echocancel=yes echotail=64 callgroup=1 devices=2 ;/etc/asterisk/extensions.conf [demo] exten = 83086921,1,Dial(CAPI/ISDN1/zzz,20,tr) exten = 83086921,2,hangup If I forward the call to a SIP phone, or start a call to my mobile phone from a SIP phone, the phones ring. Thanks for any help or hints, Ralf -- ___ Play 100s of games for FREE! http://games.mail.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 ___ --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] External IAX2 phone defined as internal behaving as from PSTN
From The CLI with iax debug (IP address faked) asterisk1*CLI Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 0ms SCall: 22774 DCall: 0 [99.99.99.212:1720] CALLING NUMBER : 1055 CALLING NAME: 1055 FORMAT : 256 CAPABILITY : 271 USERNAME: 1055 CALLED NUMBER : 1 DNID: 1 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 2ms SCall: 3 DCall: 22774 [99.99.99.212:1720] AUTHMETHODS : 3 CHALLENGE : 361908915 USERNAME: 1055 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00500ms SCall: 22774 DCall: 3 [99.99.99.212:1720] MD5 RESULT : 07051d0879d814b279c1ed58ba8a04b4 -- Accepting AUTHENTICATED call from 99.99.99.212: requested format = g729, requested prefs = (), actual format = g729, host prefs = (g729|gsm|ulaw|alaw), priority = mine Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00520ms SCall: 3 DCall: 22774 [99.99.99.212:1720] FORMAT : 256 -- Executing Macro(IAX2/1055-3, rg-group|ringall|30||1000-1007-1050-1450-1600) in new stack -- Executing Macro(IAX2/1055-3, user-callerid) in new stack -- Executing DBget(IAX2/1055-3, AMPUSER=DEVICE/1055/user) in new stack ### As we can see I jumps straight to the rg-group macro and not from-internal as expected extensions.conf [from-internal] ;allow phones to use applications include = app-userlogonoff include = app-directory include = app-dnd include = app-callforward include = app-callwaiting include = app-messagecenter include = app-calltrace include = parkedcalls include = from-internal-custom ;allow phones to dial other extensions include = ext-fax include = ext-local include = ext-group include = ext-queues include = ext-zapbarge include = ext-meetme include = ext-record include = ext-test ;allow phones to access trunks include = outbound-allroutes exten = s,1,Macro(hangupcall) exten = h,1,Macro(hangupcall) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters (Asterisk) Sent: 27 January 2006 16:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] External IAX2 phone defined as internal behaving as from PSTN On Fri, January 27, 2006 17:23, Ian Cowley said: Iax.conf [general] ;bindport = 4569 ; Port to bind to (IAX is 4569) bindport = 5036 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=g729 ; 4 simultaneous allowed allow ilbc ; prefered for iax2 allow=gsm ; 13 Kbps (full rate), 20ms frame size allow=ulaw ;(g711)64 Kbps, sample-based allow=alaw ;(g711)64 Kbps, sample-based mailboxdetail=yes jitterbuffer=yes context=from-internal #include iax_additional.conf #include iax_custom.conf iax_additional.conf [1055] username=1055 type=friend secret=# record_out=Adhoc record_in=Adhoc qualify=yes port=4569 notransfer=yes [EMAIL PROTECTED] host=dynamic context=from-internal callerid=device 1055 Regards ianC Looks like you are using AMP / [EMAIL PROTECTED] As far as I can tell, this should work correctly... There might be something going on in the translation by the Checkpoint NAT control... Have you tried iax2 debug to see what it is receiving? the first few packets should give you sufficient information... Good luck! -- 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 This Inbound email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com This email has been scanned for all viruses by the MessageLabs Email Security System. For more information on a proactive email security service working around the clock, around the globe, visit http://www.messagelabs.com ___ This e-mail
[Asterisk-Users] 802.1p
Hi, I'm trying to configure some Quality Of Service among an Asterisk server with RedHat3 and some IP phones on my LAN. I read about 802.1p (level 2) QoS, using 3 bits of VLAN tag. Two questions: - do I need to use tagged links (trunks) end-to-end? In other words, do all ports on all switches from phones to server need to be configured as 'tagged'? - how can I configure ethernet card on the Red Hat server (Broadcom, tg3 driver) to support tagged traffic and to mark outgoing packets with priority 6? Thanks in advance for any help Mimmus ___ --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: Good provider of Polycom Phones (mostly for access to latest/greatest firmware)
Hi Gavin - I've ordered a few IP501s from PC Connection, basically since we have an account with them. I like the phones for what they do, and now would like establish a relationship with a reseller that can give us maintenance and access to the most current firmware. What are some good resellers out there? I love PC Connection for most of my ordering, but I've actually never used them for Polycom hardware, even though we have a lot of it. My Polycom supplier has been http://www.tritechcoa.com They are extremely responsive, and have the best prices around on Polycom. When I recently asked for the latest firmware, I got a response back in about 2 minutes. Now I actually have firmware newer than what is listed on the Polycom website. I haven't tested it out yet, though. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AAH out bound routing problem
Hi all of them thanks for the quick reply i was tried adding 9 as well as 00 but i get number invalid if i put any of the digits what kind of config files need to post here to resolve the problem please assists ram On 1/27/06, Michael Collins [EMAIL PROTECTED] wrote: Ram, On my AAH the stock dial plan requires a 9 first. For kicks, try dialing 919197543700 and see what you get. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of ramSent: Friday, January 27, 2006 6:14 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] AAH out bound routing problem Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700 -- Got SIP response 488 Not acceptable here back from(PeerIP) -- SIP/easycall-838e is circuit-busy ram___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AAH out bound routing problem
Start with extensions.conf and also the debug lines from the Asterisk console. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ram Sent: Friday, January 27, 2006 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AAH out bound routing problem Hi all of them thanks for the quick reply i was tried adding 9 as well as 00 but i get number invalid if i put any of the digits what kind of config files need to post here to resolve the problem please assists ram On 1/27/06, Michael Collins [EMAIL PROTECTED] wrote: Ram, On my AAH the stock dial plan requires a 9 first. For kicks, try dialing 919197543700 and see what you get. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of ram Sent: Friday, January 27, 2006 6:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] AAH out bound routing problem Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700 -- Got SIP response 488 Not acceptable here back from(PeerIP) -- SIP/easycall-838e is circuit-busy ram ___ --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: Polycom 501 horrible echo
Hi - I'm running 1.6.2.0041 according to my phone. Which firmware worked for you? It was the old firmware from when we first got the phones actually. 1.4.x I think. Then I read that they fixed the CID issue and decided we needed an upgrade. I tried it out on my phone, but didn't really notice the problem until we had upgraded the rest. Oh well... Also, these are IP 500 SIP. We've been using Polycom phones since firmware version 1.3.0, and I've used every version of the firmware since then in production on IP300s, IP500s, IP501s, IP600s, IP601s and IP4000s. I've never had this issue on any of them. I don't mean to downplay the issue, but it may be possible that you did, in fact, get a bad batch of phones. When I've ordered these phones in quantity before, I've gotten many phones with consecutive serial/mac addresses, so they were probably manufactured in a bunch. Maybe a bad batch of mics got installed on a group of phones? One thing I was pondering: you are not, by chance, using the same sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has changed significantly between these versions, and certain acoustic settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention that ipmid.cfg and sip.cfg were merged in the 1.5.x release). - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AAH out bound routing problem
if you are using AAH, please post extensions.conf, extensions_additional.conf - also send us more info on your phones. thanks rajeev ram wrote: Hi all of them thanks for the quick reply i was tried adding 9 as well as 00 but i get number invalid if i put any of the digits what kind of config files need to post here to resolve the problem please assists ram On 1/27/06, *Michael Collins* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ram, On my AAH the stock dial plan requires a 9 first. For kicks, try dialing 919197543700 and see what you get. -MC *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] *On Behalf Of *ram *Sent:* Friday, January 27, 2006 6:14 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] AAH out bound routing problem Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700 -- Got SIP response 488 Not acceptable here back from (PeerIP) -- SIP/easycall-838e is circuit-busy ram ___ --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 ___ --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] Polycom Asterisk 1.2.1 and Sipura SPA3000strange problem
Hi, I've got the exact same problem here with Asterisk 1.2.1, at 2 locations. The first one have 8 sipura 3000 with 5 pstn lines and 8 standard phones. There is also 4 Mitel 5215 phones The secon one have 8 sipura 3000 with 5 pstn lines and 8 standard phones. There is also 8 Mitel 5215 phones I have the same problem as yours on Mitel or standard phones at the 2 locations. I also tried with Sipura v2 and v3 firmware... Same problem It only happen 3 or 4 times a day but it's a big pain in the ass. For the network config: At each location all the sipura are on a network of their own directly connected to the astersik box. All the mitel are on a secon network card. The asterisk machines are P4 on ASUS Motherboard in software raid1 config. Does somebody have a solution ? I've found someone else with the same problem: http://www.voipuser.org/index.php?name=PNphpBB2file=viewtopict=4207highli ght= And he also have another problem with the sipura that I have too. http://www.voipuser.org/index.php?name=PNphpBB2file=viewtopict=4193highli ght= Thanks for your help... ___ Jean-François Rousseau www.sys-tech.net [EMAIL PROTECTED] Tél. 24h (418) 520-0739Télec. (418) 520-4554 1-877-969-tech Ouverture Technologique -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de C F Envoyé : 28 décembre 2005 15:15 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000strange problem In any case I'm trying to figure out if maybe someone else has seen this problem. Or if they know what it might be. On 12/28/05, C F [EMAIL PROTECTED] wrote: For somereason I think it's the polycom, which means I need logging for the Polycom and not the spa. On 12/28/05, Rich Adamson [EMAIL PROTECTED] wrote: I have the follwoing setup: Asterisk SVN-tag-1.2.1-r7367 6 Polycom 500 Sip version 1.5.x 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat single network, no NAT, and no gateways to reach each other. Sometimes (happens around 3 times a day, but sometimes far more often), while on the phone to an outside caller (on the PSTN using the FXO on the spa3k), the call dissconects from the polycom and goes thru the incoming extension for the sipura. In other words, astrisk at least as far as I can see from what gets executed in the DP (and maybe spa3k) sees this as if the follwoing has happened: 1. The polycom user hungup, 2. A new call came in on the spa3k. The follwoing is part of the log that I think might help: Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from channel: SIP/201-8ba1 Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels SIP/201-8ba1 and SIP/804-fd83 SIP/201 is the Polycom, while SIP/804 is the spa3k. If I'm losing a frame, is there a way to configure asterisk not to drop the channel? Or is this something the Polycom/Sipura are doing? FYI, asterisk is running on a VIA/MPIA platform. Pure guess is that something happened (unknown what) and the error messages posted above are the result of that, and not the root cause. Finding the root cause may require you to implement the syslog server and debug server options in the spa3k, and compare those log entries to what * records for log messages during a failure. Implementing the log functions on the spa3k does require a reboot. Their log messages are rather cryptic, but looking at keywords and timestamps might identify which box(es) are involved with the dropped calls. ___ --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] AAH out bound routing problem
Hi rajeev
i have posted the extension.conf before
now iam posting extension_additional.conf
[EMAIL PROTECTED] asterisk]# more extensions_additional.conf[globals]#include globals_custom.confVM_PREFIX = *RINGTIMER = 15REGTIME = 7:55-17:05REGDAYS = mon-friRECORDEXTEN =
PARKNOTIFY = SIP/200OUT_2 = SIP/easycallOUTPREFIX_2 =OUTMAXCHANS_2 = 1
OUTCID_2 = outside account
OPERATOR =NULL = IN_OVERRIDE = forcereghoursINCOMING = group-allFAX_RX_EMAIL = [EMAIL PROTECTED]FAX_RX = systemFAX =DIRECTORY_OPTS =DIRECTORY = last
DIAL_OUT = 9DIAL_OPTIONS = trDIALOUTIDS = 2/CALLFILENAME = AFTER_INCOMING =
[ext-local]include = ext-local-customexten = 1000,1,Macro(exten-vm,1000,1000)exten = ${VM_PREFIX}1000,1,Macro(vm,1000)exten = 1000,hint,SIP/1000
[outbound-allroutes]include = outbound-allroutes-custominclude = outrt-001-longdistance
[outrt-001-longdistance]include = outrt-001-longdistance-customexten = _1NXXNXX,1,Macro(dialout-trunk,2,${EXTEN},)exten = _1NXXNXX,2,Macro(outisbusy) ; No available circuitsexten = _NXXNXX,1,Macro(dialout-trunk,2,${EXTEN},)
exten = _NXXNXX,2,Macro(outisbusy) ; No available circuitsexten = _NXX,1,Macro(dialout-trunk,2,${EXTEN},)exten = _NXX,2,Macro(outisbusy) ; No available circuits
ram
On 1/27/06, Rajeev Natarajan [EMAIL PROTECTED] wrote:
if you are using AAH, please post extensions.conf,extensions_additional.conf - also send us more info on your phones.
thanksrajeevram wrote: Hi all of them thanks for the quick reply i was tried adding 9 as well as 00 but i get number invalid if i put any of the digits
what kind of config files need to post here to resolve the problem please assists ram On 1/27/06, *Michael Collins*
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Ram, On my AAH the stock dial plan requires a 9 first.For kicks, try
dialing 919197543700 and see what you get. -MC *From:*
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
] *On Behalf Of *ram *Sent:* Friday, January 27, 2006 6:14 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] AAH out bound routing problem
Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and
http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700
i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage
any help Called easycall/19197543700 -- Got SIP response 488 Not acceptable here back from (PeerIP) -- SIP/easycall-838e is circuit-busy
ram ___ --Bandwidth and Colocation provided by Easynews.com
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RE: [Asterisk-Users] extension to extension dialing
Hmm.. I definitely have type=friend in the sip.conf and I added
qualify=yes but, I think that's default anyways.. When I call from the
outside and enter his extension it goes through to him fine but, when I
go extension to extension it automatically goes to voicemail.. Here are
the messages from the console:
-- Executing Macro(SIP/130-58df, stdexten|SIP/124) in new stack
-- Executing Dial(SIP/130-58df, SIP/124|20) in new stack
-- Called 124
Jan 27 10:27:10 WARNING[28243]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Critical Request)
== No one is available to answer at this time
-- Executing Goto(SIP/130-58df, s-NOANSWER|1) in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing VoiceMail(SIP/130-58df, u124) in new stack
-- Playing 'voicemail/default/124/greet' (language 'en')
Jan 27 10:27:10 NOTICE[28243]: sched.c:290 ast_sched_del: Attempted to
delete non-existant schedule entry 22838!
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
Richardson
Sent: Thursday, January 26, 2006 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extension to extension dialing
In your sip.conf, make sure these phones have a Type=Friend entry and
a qualify=yes. I don't think the qualify=yes is required, but it helps
in debuging.
About the port, I'm not too sure about sipura and snom phones (I only
have Cisco phones :(). That could have something to do with it..
On 1/26/06, Nora Lavelle [EMAIL PROTECTED] wrote:
Hi there gary. thanks so much for your help. we're using sipura-841
and snom 320s.
Here's the sip show peers.. that's weird that extension 130 has port
2057.. could that be the problem ?
-nora
Name/usernameHostDyn Nat ACL Mask Port
Status
201/201 10.200.0.56 D 255.255.255.255 5060
Unmonitor
ed
130/130 10.200.0.10 D 255.255.255.255 2057
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129/129 10.200.0.5 D 255.255.255.255 5060
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127/127 10.201.0.30 D 255.255.255.255 5060
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126/126 10.201.0.29 D 255.255.255.255 5060
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125/125 10.201.0.35 D 255.255.255.255 5060
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124/124 10.201.0.31 D 255.255.255.255 5060
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102/102 10.200.0.48 D 255.255.255.255 5060
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-Original Message-
From: [EMAIL PROTECTED] on behalf of Gary
Richardson
Sent: Thu 1/26/2006 5:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extension to extension dialing
Check your error messages in you asterisk console. Perhaps your sip
secret or caller id is broken?
What type of phones are you using?
On 1/26/06, Nora Lavelle [EMAIL PROTECTED] wrote:
Sorry for all the newbie questions. I really appreciate everyone's
help
today.
Okay I've got outgoing and incoming calls working with no echo. yay!
Now I'm
having an issue with SIP extension to extension calling. Any time I
dial
another extension it goes right into voice mail. My extensions.conf
is
pretty small and rough but, here's what I have right now. Most of it
was
taken from the voip-info website. Any help as always VERY
appreciated.
Thanks again!
Nora Lavelle
# cat extensions.conf
[incoming]
exten = s,1,Answer();
exten = s,2,Background(ssn-greeting);
exten = *,1,Directory(default)
exten = 205,1,Wait(2)
exten = 205,2,Record(/tmp/asterisk-recording:gsm)
exten = 205,3,Wait(2)
exten = 205,4,Playback(/tmp/asterisk-recording)
exten = 205,5,Wait(2)
exten = 205,6,Hangup
[internal]
exten = 101,1,Macro(stdexten,SIP/101)
exten = 102,1,Macro(stdexten,SIP/102)
exten = 103,1,Macro(stdexten,SIP/103)
exten = 123,1,Macro(stdexten,SIP/123)
exten = 124,1,Macro(stdexten,SIP/124)
exten = 125,1,Macro(stdexten,SIP/125)
exten = 126,1,Macro(stdexten,SIP/126)
exten = 127,1,Macro(stdexten,SIP/127)
exten = 128,1,Macro(stdexten,SIP/128)
exten = 129,1,Macro(stdexten,SIP/129)
exten = 130,1,Macro(stdexten,SIP/130)
exten = 135,1,Macro(stdexten,SIP/135)
exten = 117,1,Macro(stdexten,SIP/117)
exten = 201,1,Macro(stdexten,SIP/201)
[voicemail]
exten = 300,1,Answer
exten = 300,2,VoicemailMain(ssn-voicemail-greeting)
exten = 300,3,Hangup
[local]
exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _9NXX,2,Congestion
[longdistance]
exten = _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _91NXXNXX,2,Congestion
[macro-stdexten]
exten = s,1,Dial(${ARG1},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten =
RE: [Asterisk-Users] Re: Random Disconnects
Hi, we have the same problem here at 2 location that we just installed Asterisk 1.2.1 P4 3.0Ghz Motherboard ASUS P4S800-VM 2 SATA disk in software Raid-1 We use 2 nic, one (onboard) to talk to the network (1Gbps link that we use à 100Mbps) and the other realtek 8139 from Startek that talk to the sipura on a separate subnet. Up to now I've tried going back to asterisk 1.0.9 with no success Tried V2xx and V3xx of the sipura without success Have you found something ? Thanks in advance ___ Jean-François Rousseau www.sys-tech.net [EMAIL PROTECTED] Tél. 24h (418) 520-0739Télec. (418) 520-4554 1-877-969-tech Ouverture Technologique -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Thczv F. Thczv Envoyé : 26 janvier 2006 14:12 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Re: Random Disconnects On 1/26/06, C F [EMAIL PROTECTED] wrote: OK, some update on this. It's not related to the Sipuras (actualy the sipuras are very good at this, since they will re-ring your call). I changed my setup to a mediatrix 1204 and I still have the problem. Right now I'm looking at: 1. Changing the NIC. 2. Changing the machine asterisk is on. I will start with one, if that fails, then I'm going with a new machine (such fun:P) BTW, what NIC are you using? what chipset is it? what module makes it work? and/or what option in the kernle did you compile that loads it? A 'dmesg | grep eth' should give you some info. I believe the NIC in the asterisk machine is a Netgear FA310TX. I really didn't do anything manually as part of the compile. The [EMAIL PROTECTED] CD took care of that for me (though I stripped out sip.conf and extensions.conf and configured those myself). Here is what dmesg | grep eth returns: * divert: allocating divert_blk for eth0 eth0: Lite-On 82c168 PNIC rev 32 at 0xc48db000, 00:A0:CC:D6:A9:47, IRQ 3. divert: freeing divert_blk for eth0 divert: allocating divert_blk for eth0 eth0: Lite-On 82c168 PNIC rev 32 at 0xc496f000, 00:A0:CC:D6:A9:47, IRQ 3. eth0: Setting full-duplex based on MII#1 link partner capability of 45e1. * Dave ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.3 CentOS 4.x RPMS
Eric Bishop wrote: Do you have step by step instructions on how you created these RPMs. I would like to create a few of my own but compiled for my own custom kernel and patchea and am not very familiar with RPM packaging A good starting point is to download and install the source RPMs in: ftp://ftp.linuxsys.com/pub/releases/CentOS-4.0/asterisk-1.2.3/SRPMS Install them and then tweak the spec file in '/usr/src/redhat/SPECS/' and do a rpmbuild on it. Rod -- ___ --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: Polycom 501 horrible echo
I have had no problems running the Sip.cfg from 1.5.2 with 1.6.4 so far, but I am looking to update in the next while. Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: January 27, 2006 1:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Polycom 501 horrible echo Hi - I'm running 1.6.2.0041 according to my phone. Which firmware worked for you? It was the old firmware from when we first got the phones actually. 1.4.x I think. Then I read that they fixed the CID issue and decided we needed an upgrade. I tried it out on my phone, but didn't really notice the problem until we had upgraded the rest. Oh well... Also, these are IP 500 SIP. We've been using Polycom phones since firmware version 1.3.0, and I've used every version of the firmware since then in production on IP300s, IP500s, IP501s, IP600s, IP601s and IP4000s. I've never had this issue on any of them. I don't mean to downplay the issue, but it may be possible that you did, in fact, get a bad batch of phones. When I've ordered these phones in quantity before, I've gotten many phones with consecutive serial/mac addresses, so they were probably manufactured in a bunch. Maybe a bad batch of mics got installed on a group of phones? One thing I was pondering: you are not, by chance, using the same sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has changed significantly between these versions, and certain acoustic settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention that ipmid.cfg and sip.cfg were merged in the 1.5.x release). - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_bluetooth: successful compile and outbound cell calls: Still tweaking inbound setup. WAS: Cannot compile chan_bluetooth on Asterisk 1.2.1
Editing subject line to reflect current status. On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote: Since T616 is not answering (and incoming calls are going to Cingular voicemail after 30 sec,) I suspect the problem focus area is... -- Executing Answer(BLT/T616, ) in new stack Is http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz tar xzf bluetoothfiles.tar.gz the latest source (r40?) On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote: Here are my findings with my experiment using Sony Erisson T616 with Cingular Service and connected to [EMAIL PROTECTED] 2.2 on a freshly installed system and following the instructions http://www.thetechguide.com/howto/asterisk/chanbluetooth.html Outbound calls (Asterisk to T616 via bluetooth): Works OK via Dial(BLT/T616/8005551212) Inbound calling (T616 to asterisk via bluetooth): My configuration for inbound calls: [bluetooth] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,Dial(SIP/1007,15,rtT) exten = s,4,VoiceMail([EMAIL PROTECTED]) exten = s,5,Hangup My observation: When I call my cell T616 from my landline, SIP/1007 rings for 2 seconds and the call is answered by Cingular voicemail not by asterisk voicemail. My cingular voicemail is set to answer in 30 seconds after first ring. Output on the asterisk CLI: [EMAIL PROTECTED] ~]# asterisk -r Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk 1.2.1 currently running on asterisk1 (pid = 3025) Verbosity is at least 3 [AG] T616 +CIEV: 2,4 [AG] T616 +CIEV: 2,3 [AG] T616 RING [AG] T616 +CLIP: 421212,161,,,Landline -- Executing Wait(BLT/T616, 1) in new stack -- Executing Answer(BLT/T616, ) in new stack [AG] T616 +CIEV: 2,1 [AG] T616 +CIEV: 3,0 -- Executing Dial(BLT/T616, SIP/1007|15|rtT) in new stack -- Called 1007 -- SIP/1007-d97e is ringing == Spawn extension (bluetooth, s, 3) exited non-zero on 'BLT/T616' [AG] T616 ATH [AG] T616 AT+CHUP [AG] T616 ERROR [AG] T616 OK [AG] T616 AT+BRSF=23 [AG] T616 ERROR [AG] T616 AT+CIND=? [AG] T616 +CIND: (battchg,(0-5)),(signal,(0-5)),(batterywarning,(0-1)),(chargerconnected,(0-1)),(service,(0-1)),(sounder,(0-1)),(message,(0-1)),(call,(0-1)),(roam,(0-1)),(smsfull,(0-1)) [AG] T616 OK [AG] T616 AT+CIND? [AG] T616 +CIND: 5,3,0,1,1,0,0,0,0,0 [AG] T616 OK [AG] T616 AT+CMER=3,0,0,1 [AG] T616 OK [AG] T616 AT+CLIP=1 [AG] T616 OK [AG] T616 AT+CGMI [AG] T616 SONY ERICSSON [AG] T616 OK [AG] T616 AT+CGMI [AG] T616 SONY ERICSSON [AG] T616 OK [AG] T616 +CIEV: 2,4 [AG] T616 +CIEV: 2,3 asterisk1*CLI On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote: BTW, I did get clear bidirectional audio when I succeded in dialing out...(with the channel = 3 in /etc/asterisk/bluetooth.conf) I have Sony Ericsson T616 connected to a cheap commodity bluetooth USB dongle that I bought ages ago from meritline. On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote: Thanks a billion. Outbound bluetooth dialling on the lines of Dial(BLT/DevName/8005551212) worked for me. Still trying out the inbound route. Before I created the [bluetooth] context, it tried to reach the [default] context but then I began by creating a new context [bluetooth] in extensions.conf and got my internal SIP phone to ring when I received a call on my SE T616 cell phone. However, I could not get the inbound line answered and I will continue to work on this over the weekend and report back my progress. On 1/25/06, Joseph Tanner [EMAIL PROTECTED] wrote: Again, my documentation is still sparse. I should have noted that the phone will recognize asterisk and connect even if the channel in bluetooth.conf is configured wrong. You'll just get no audio, or disconnects, or what-not until it's set correctly. So realize that later on when you're testing. Also the usb dongle must have a CSR chipset, else it won't work (well, at least probably won't work, I'll provide instructions on how to tell if it should work or not later). Here's the relevant instructions on http://www.crazygreek.co.uk/content/chan_bluetooth for how to dial out: Send a call out by using Dial(BLT/DevName/0123456). As far as dialing in, there's a special context (I think [bluetooth] maybe? I'll have to get back to you on that). I know that it should work fine, because I tried dialing the phone, asterisk picked it up then immediately disconnected because there was no context for
[Asterisk-Users] Nagios and Asterisk
Is anyone using Asterisk (and Festival) to make calls to appropriate persons (techs, etc. ) when Nagios generates a particular type of alert? If so, I would love to hear how people are doing it. Thanks, -- Darrell S. Long BestWeb Corporation ___ --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 channel not diconnecting on hangup
I've got an SPA-841 SIP hardphone connecting to my asterisk server across the internet through a couple of NAT routers. Everything works great (I can initiate calls, receive calls, hear audio both ways, etc.) except for one thing. When I hang up the phone, the connection in asterisk doesn't disconnect. The phone is idle and things everything is fine, but Asterisk still show an open channel. It's like the phone isn't sending some sort of disconnect message to Asterisk. Can anyone provide some ideas on what might be going wrong? As a test case, I call my echo() extension from the remote phone. The connection works fine but when I hangup the phone and get information from the Asterisk console here's what I see: [Jan 27 10:27:00] -- Executing Playback(SIP/scottbhome-f4de, demo-echotest) in new stack [Jan 27 10:27:00] -- Playing 'demo-echotest' (language 'en') [Jan 27 10:27:19] -- Executing Echo(SIP/scottbhome-f4de, ) in new stack I hangup the phone here pbx*CLI show channels Channel Location State Application(Data) SIP/scottbhome-f4de [EMAIL PROTECTED]:2 Up Echo() 1 active channels 1 active calls pbx*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message xx.xx.xx.xxscottbhome 304dcbc8-5f 00101/00102 g729 No Rx: INVITE 1 active SIP channels So the connection initiates correctly, but nothing ever terminates it. I finally do a SOFT HANGUP to kill the connection. Thanks for any help! ___ --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: Good provider of Polycom Phones (mostly for access to latest/greatest firmware)
amen! http://www.tritechcoa.com/ is a great supplier :) Noah Miller wrote: Hi Gavin - I've ordered a few IP501s from PC Connection, basically since we have an account with them. I like the phones for what they do, and now would like establish a relationship with a reseller that can give us maintenance and access to the most current firmware. What are some good resellers out there? I love PC Connection for most of my ordering, but I've actually never used them for Polycom hardware, even though we have a lot of it. My Polycom supplier has been http://www.tritechcoa.com They are extremely responsive, and have the best prices around on Polycom. When I recently asked for the latest firmware, I got a response back in about 2 minutes. Now I actually have firmware newer than what is listed on the Polycom website. I haven't tested it out yet, though. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- 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] Nagios and Asterisk
I have used both, just not together. I have a possible idea though. If they're running on separate servers, you can have nagios send an email that the asterisk server receives. Have different email aliases for different alerts, or have a script parse the email to see what kind of alert it is. Have this script generate a .call file in /var/spool/asterisk/outgoing based on the type of alert. If they're running on the same server you might be able to skip having to send an email (but if not, then just have it send an email to a local user, it'll work the same). Personally, I just had Nagios send an email whenever there was a problem. If the tech is in front of their workstation, they'll get a notice immediately. If not, you could have a text message sent instead. Worked great for me. On 1/27/06, Darrell Long [EMAIL PROTECTED] wrote: Is anyone using Asterisk (and Festival) to make calls to appropriate persons (techs, etc. ) when Nagios generates a particular type of alert? If so, I would love to hear how people are doing it. Thanks, -- Darrell S. Long BestWeb Corporation ___ --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] Linksys SPA-941 multiple line appearences
Accounts 3-4 are disabled, Account 1 is the only account. Thats it. Nothing special. If you have problems, try doing *70. -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tracinetSent: Wednesday, January 25, 2006 8:57 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Linksys SPA-941 multiple line appearences This may sound odd, but I have the opposite problem. I am trying to disable call waiting by only allowing 1 call to come into the SPA-941, but I am getting 2 calls per line key. If have 1 extension set up on all 4 line keys, the phone handles 8 incoming calls. I have a few customers that like that feature, but others that want to just have 1 call - any tips on what you did to only get 1 call to ring through without doing any status checking in the asterisk dialplan would be appreciated.- Pedro On 1/25/06, Michael Keyes [EMAIL PROTECTED] wrote: I set up only 1 extension and set all 4 line appearences to point to thatextension.I could place up to 4 outgoing calls as extension 1 no problem.The problem happened when there was an active call on line appearence 1 and someone called extension 1.Instead of ringing on line appearence 2(extension 1) it would busy out.My question, is there something inAsterisk that needs to be adjusted to make the phone work properly or isthere a setting on the phone that needs to be adjusted?Thank you.Michael K-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of KerryGarrisonSent: Tuesday, January 24, 2006 10:33 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Linksys SPA-941 multiple line appearencesYou only need to setup ONE account and all four call appearances will work.-Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Keyes Sent: Tuesday, January 24, 2006 3:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Linksys SPA-941 multiple line appearences Has anyone had any experience with the Linksys SPA-941 when it comes to multiple line appearences? The 941 comes with 4 line appearence buttons which can individualy be configured to point at any extension.The phone is capable of 2 extensions out of the box with the option to add2 more for a license fee. The 941 manual states that, "The Call Waiting function is activated when a device has a call in the active state and another call is incoming. The phones in the SPA series do not support multiple calls on the same Line Key. Incoming calls are assigned to an unused Line Key, causing the Line Key to quickly blink red. (Note that the Voice Mail Waiting Indicator also blinks red whenever there is an incoming call.) The phone will not ring. However, to alert the user, the call waiting tone is played into the active audio device." During testing I set up an Asterisk 1.2 box with a 941 phone using firmware ver. 4.1.8.I configured 1 extension and set all 4 line appearence buttons to point to that extension.If there was an active call in progress I could place that call on hold and by pressing line appearence button 2 was able to place an outgoing call.That outgoing call would appear to come from extension 1.This is all working as desired. If an active call was in progress and someone called my extension the product manual indicates that call should appear on line appearence button 2.During my testing Asterisk would flag my extension 1 as busy and instead of ringing the phone on line appearence 2 would send the call to voicemail. Is anyone aware of any configuration setting needed on either the Asterisk server or on the phone to make Call Waiting function as described in the manual?Thank you. Michael K ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service
I tried through voipdiscount as well. Even my older account through voipbuster started to behave this way and it used to be ok on IAX. I would expect at least some reply. Rudolf - Original Message - From: Aryanto Rachmad [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 26, 2006 5:00 PM Subject: Re: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service Didn't you read this from their QA? I want to configure my own IAX/SIP device for calling with VoipBuster, is that possible? It is possible to use your own IAX/SIP device, however we do not support it. We advise you to use SIP-Discount instead. Do you have the same problem when you use their softphone? If not, why complaining. The call to the UK is free only for VoIPstunt - Original Message - From: RumaTech [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 26, 2006 6:19 AM Subject: [Asterisk-Users] Voipbuster/voipstunt -- what a crap service Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20 02:32:43 WARNING[4853]: chan_sip.c:8520 handle_response: Got authentication request (401) on unknown BYE to 'sip:[EMAIL PROTECTED];tag=c9ebef50c90078c2c93eddc243d7352d6e04' Secondly, they charged me for calls to UK that was supposed to be free. And their customer service does not respond at all. Do they have a phone number I can call? Thanks, Rudolf ___ --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
[Asterisk-Users] Lockups since upgrade 1.2.3 - anyone else? Any ideas?
Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 hours or so. Since upgrading to 1.2.3, though, the whole system has locked up twice. Once on Thursday, and then about a half hour ago. The server would reply to a ping, but no ssh login, no local console login - just locked up. This ain't good for business. I have scoured the logs, and find no errors. Not even right before/around the time of the crash. I am worried that 1.2.3 is not as stable as 1.0.9 (or 1.0.10, though we never ran that version). Is there a needed step aside from make; make install that I missed when upgrading? Has anyone else had similar problems? Or, if I submit other info, would someone have a clue as to what to look at? We run a TDM400P with 3 FXO modules, and about 15 SIP Cisco 79XX phones here. Any help is appreciated, this cannot continue. Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.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
