Re: [Asterisk-Users] ILBC to G711 transcoding experince ?
Thank You for replying. There is some progress, and confusion, so i posted the 2nd post about it. Here is what is happening. When call comes from Machine A to Machine B, and Machine B is running asterisk 1.0.9 the voice on ilbc is fine, i played mp3 on it. But when the asterisk on machine B is 1.2.x there is no voice on machine a coming from machine B. I was able to send the call using 1.0.9 to machine C however, the problem now is that the call going to machine C has incorrect user name It should be [EMAIL PROTECTED] its going as [EMAIL PROTECTED] using 1.0.9 With 1.2 user name is going correctly There are no firewalls or other issue so far. It seems like an RTP issue, but i don't know what to fix. My msn is [EMAIL PROTECTED] if we can continue there. Rehan it shouldnt be a problem from ILBC to g711u/a , but for g729 you need a licence, otherwise no transcoding can ocurr. However does not seems to be your problem, since the call should be hanged up, and you just dont receive audio. That seems to me more like a problem with RTP not finding a right route to the final UA. Can you be a little more specific about where are located machines A,B,C. Firewalls, etc? regards On 1/13/06, Rehan AllahWala [EMAIL PROTECTED] wrote: Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already tested: Xpro Logged in on Machine B using ILBC sending to Machine C and it works fine. Do send me your charges. Thank You, Rehan Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --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] Echo
Just checking ___ --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] Adit 600 and echo
-Original Message- From: Gregory Wiktor - ADCom Corp. Sent: Sunday, January 15, 2006 3:29 PM To: 'Patrick' Subject: RE: [Fwd: RE: [Asterisk-Users] Adit 600 and echo] Hello Patrick, I believe mine is the 2572. 64ms EC. Power supply: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=7574802048 Yes got them off ebay, but keep your eye open. I just checked, I got the EC for only 30.49 including shipping from ebay :) http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5824630277 Yay... I'll try to take some pictures this week for the wiki... The soldering is easy but it will require good quiet time. Then I ran the lines 2 a 2 port cat6 block. Greg -Original Message- From: Patrick [mailto:[EMAIL PROTECTED] Sent: Sunday, January 15, 2006 10:22 AM To: Gregory Wiktor - ADCom Corp. Subject: [Fwd: RE: [Asterisk-Users] Adit 600 and echo] Forwarded Message From: [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Adit 600 and echo Date: Sun, 15 Jan 2006 03:50:10 -0500 See my other post. I had the exact same problem on about 5 lines. I got a tellabs ec a while back, and then a power supply but hadn't the time to solder it up and get things going...I had just been forwarding the inbound calls to voicepulse connect. I finally did get it soldered and ready, and when I installed the new hardware tellabs ec, echo was gone. I must have spent 10-20 hours on this problem, and the tellabs solved it in 5 minutes (once setting the channel properties to FXO-LS FXS-LS) I think it cost something like $110 for the card, and $86 for the power supply. Then you need the time to do the soldering. Greg, Can you please tell me the modelnumber of the Tellabs' card and power supply and any info/links regarding the soldering? Where did you buy the card and power supply, eBay? Thanks and regards, Patrick ___ --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] tuning an x100p in Australia for echocancellation
James Harper wrote on Saturday, 14 January 2006 3:17 PM: Are the AU telephone standards the same as US standards (eg, 600 ohm impedence)? This is a question I've been trying to answer too. I had a look at the standard phone that Telstra would provide to customers about 5 years ago, and it has an impedence switch on the bottom to toggle between 'NORM' and '600', which suggests that 600 ohms isn't the normal impedence. On an au configuration example for the pap2 I have seen on the web, the impedence is set to '220+820||120nF', which suggests that our standard here isn't 600. Indeed, the AU standard is just what you quoted. It is complex impedance. A 220-ohm resistive load connected in series to an 820-ohm resistive load which is connected in parallel with a 120nF cap. Does anyone know of an addon device which can do impedence matching on the line, or of a modification to the card I don't know if this will help you, but you may do a Google search for the ETAL P3324 and P3356. I believe they may handle the impedance matching you need. Here is a link to a PDF spec sheets: http://www.ibselectronics.com/pdf/pa/etal/line_P3324.pdf http://www.ibselectronics.com/pdf/pa/etal/line_P3356.pdf -- Trevor Hammonds ___ --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 with Cisco
Hi, Does anyone have any real world examples of setting up Asterisk to break out to the PSTN via a Cisco router. I have a 2801 with a PVDM2-8 and -1MFT-E1 connected to a ISDN30 PRI circuit. Is it possible to get Asterisk to talk to the Cisco Router, and what is the best protocol to use. I understand the Cisco talks h232 or SIP, but am unsure as the best way to do this. If anyone has any pointers Id be grateful :) Cheers, Jo ___ --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] Test to see if I'm still on list...
As I haven't received any posts since yesterday... ___ --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] PrimuX Cards with chan_capi-cm
Here is the backtrace outgoing via german NTBA : (gdb) backtrace #0 0x4202d5e7 in memcpy () from /lib/tls/libc.so.6 #1 0xb7519b6c in capi_handle_data_b3_indication (CMSG=0x810d12a, PLCI=158, NCCI=158, i=0x810c420) at chan_capi.c:2688 #2 0xb75162e1 in capi_handle_msg (CMSG=0xb74cc2b0) at chan_capi.c:3423 #3 0xb7513954 in do_monitor (data=0x0) at chan_capi.c:4113 #4 0x42525ccd in start_thread () from /lib/tls/libpthread.so.0 #5 0x4208fb0e in clone () from /lib/tls/libc.so.6 Am Samstag, den 14.01.2006, 15:20 +0100 schrieb Armin Schindler: On Fri, 13 Jan 2006, Christian Peter wrote: Hello List, I'm trying to get a PrimuX Card (www.primuxisdn.de) working. The Manufacturer says that chan_capi (the older one) used to work. Now I'm trying with chan_capi-cm and have got the following problems: Outgoing calls ausgehend_ueber_ntba_primux.txt the other phone rings but when I answer the call asterisk crashes. Incoming: first tried with immediate=no eingehend_von_ntba_primux_immediate_no.txt. This seems to be the same behaviour as someone on this list had this week. So I tried with immediate=yes eingehend_von_ntba_primux.txt. There is no ringing on my sip phone and chan-capi-cm says capicard_primux_2_1: too much voice to send for NCCI=0x10101 Any hints? I cannot find where the seg-fault happens. Can you create a backtrace from coredump? Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Oooh / ahhh . . . 5 tellabs boards on ebay.
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5854671489ssPageName =MERC_VIC_ReBay_Pr4_PcY_BIN_Stores_IT#ebayphotohosting Worth considering for some . . . :) I got my unit from the same fellow, worked out fine... Greg ___ --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: ignore me, just a test
sorry, just a test, seems I'm no more receiving mails ... ___ --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] Zapata.conf Realtime?
The text in extconfig.conf leads me to believe that this file can be configured from a database as well. Has anyone managed to get this working??? Thanks! Steve ___ --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] CVS HEAD chanisavail not working for sip channel?
Dear all,
I have encountered problem with app_chanisavail for sip
channels. I have setup call-limit=1 in sip.conf as instructed, but when
making call to app_chanisavail, the channels did not increment correctly. I
end up dialing multiple times to the first channel only. I think the
ast_device_state(trychan) did not returned correctly. Any idea?
Extensions.conf :
exten = _1234.,1,ChanIsAvail(SIP/1SIP/2SIP/3SIP/4SIP/5|s)
exten = _1234.,n,Dial(${AVAILORIGCHAN}/0${EXTEN:4}|45)
exten = _1234.,n+101,busy
sip.conf :
[1]
type=friend
context=default
host=xxx.xxx.xxx.xxx
username=abcd
secret=abcd
port=5060
call-limit=1
fromuser=abcd
fromdomain=xxx.xxx.xxx.xxx
nat=yes
canreinvite=no
insecure=yes
insecure=very
disallow=all
allow=g723
allow=g729
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[Asterisk-Users] new in asterisk world
Hi, I'm new in asterisk world. I have questions. For example I have my server with public IP address, but two customer with softphone in a private network. How can I do to make them work with the asterisk server? Best Regards -- Ever Zalazar ___ --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] Advice Of Charge (AOC) ?
On Sunday, January 15, 2006 10:32 AM Armin Schindler wrote: It is not implemented in chan_capi yet, but this is very easy. The question is what should be done with the AOC information? Just set some variable? As far as I know Asterisk has no API/structure for that. At least none that would help you. See http://bugs.digium.com/view.php?id=6152 and put your requests there please. There are a few things that need to be done: 1. Ensure that AOC values survive channel changes. 2. Ensure that AOC information can be retrieved/saved in the dialplan/cdr. 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] 32 E1's in one Asterisk 'box'
On Wed, 11 Jan 2006, [EMAIL PROTECTED] wrote: hi, Thanks - I was hoping someone who had done this would pop-in. Do you treat each Asterisk server as a separate entity or do you have a sentralized Asterisk that perform call-control for all etc? How do you make them behave as one, or is this not needed? Also, do you switch voice from B-channel's on one server to the B-channel's on another? In case how do you do this? SIP w/rtp/rtcp, TDMoE or ? Do you have any measurement of latency etc? Hi Jan, This site is a specialised outbound call centre (/me ducks). They all behave as one because the custom dialler application that runs on each server all talk to a single core database. Similarly, we don't register agents/SIP phones using SIP registers. Instead, their presence is captured in the database. When a server has a call and needs an agent it looks for the next free in the central database. We don't switch calls from PRI on one box to another. If I did, I would surely use an IAX2 link. Latency has never been an issue and I've never measured it. Rest assured that IAX2 on an Ethernet adds very little latency - I'd guess 20msec packetization delay, the wire latency (5ms). So unless you have a special requirement you certainly won't make a latency that humans will notice. Steve ___ --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] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
Ben Fried wrote: On 1/9/06, Rich Adamson [EMAIL PROTECTED] wrote: Sorry in advance if this is a FAQ... I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have a TDM400 card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM card. I haven't been able to get inbound fax with spandsp and rxfax to work. Occasionally an all-text fax will come in, though it's usually badly corrupted, but in most cases, it would appear that the call is terminated without successful transmission of the fax. I get logs that look what's included below. From reading the list, it looks like this is caused by the TDM card missing frames. Does that sound correct? If so, is there any relief in sight? Its been a problem since the card came out a couple of years ago. So, no it does not appear there is any relief in sight. Sigh. What a disappointment! Are there any other options for home users to receive faxes over the PSTN through *? Is anyone working on an alternative to the zaptel driver that might fix this issue? Humm, I tried to get my TDM400 card accepting faxes last week. It works about 1 out of 8 times. When it works, it looks great. When it doesn't, I usually (but not always) get a 'poor line quality' error from the sending fax machine and a blank or small corrupt image. I've tried adjusting the gain up and down, reversing ring/tip, and a few other little things. I wonder if it helps to adjust some other settings in the zapata.conf, like echo cancellation? My hope is dwindling, though, after reading this thread. :'( I do have a couple unused X100P clones sitting in that box that might be worth a try... 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] a2blling billing system
Hi i have checked with this URL but still iam not able to integrated does any one successfully integrated here ram On 1/12/06, Samy Antoun [EMAIL PROTECTED] wrote: --- Tomá¹ Komárek [EMAIL PROTECTED] wrote: Hello, I am trying to setup a2billing system for asterisk. I have installedit corectly, but I have not found any users manual. I do not understandthe whole structure. How do the parts like calling cards and sip friends cooperate together?Tomas,The calling card application can work in 2 ways:1. Customer calls a number (At you * server), the system will promptfor a pin number, after authenticating, a balance is anounced, customer is ready to make a call.2. Customer will use a SIP or IAX phone (Software or Hardware) toregister, in this option you need to create a SIP or IAX Friend.I've created a page on how to install it, but it's based on [EMAIL PROTECTED]:http://samyantoun.50webs.com/asterisk/athome/a2billing/__ Do You Yahoo!?Tired of spam?Yahoo! Mail has the best spam protection aroundhttp://mail.yahoo.com___--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] Advice Of Charge (AOC) ?
I'm very surprised that Asterisk (PBX !) do not support AOC. Setting some variable with AOC informations should be enough. Storing AOC in CDR would be perfect. P. - Original Message - From: Armin Schindler [EMAIL PROTECTED] On Sun, 15 Jan 2006, Pisac wrote: Do Asterisk support Advice Of Charge (AOC) on ISDN lines? Do any ISDN drivers (bristuff, capi, vISDN, mISDN) support AOC? It is not implemented in chan_capi yet, but this is very easy. The question is what should be done with the AOC information? Just set some variable? As far as I know Asterisk has no API/structure for that. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] realtime voicemail
i tried to setup realtime voicemail recently with 1.2.1 but couldn't get it to work. no matter what i do. it still looks for config in the voicemail.conf file. (BTW realtime sip extensions works fine) here's the voicemail line in extconfig.conf: voicemail = mysql,asterisk,voicemail here's the mysql schema: CREATE TABLE voicemail ( uniqueid int(11) NOT NULL auto_increment, customer_id bigint NOT NULL default '0', context varchar(50) NOT NULL default '', mailbox bigint NOT NULL default '0', password varchar(10) NOT NULL default '0', fullname varchar(50) NOT NULL default '', email varchar(50) NOT NULL default '', pager varchar(50) NOT NULL default '', stamp timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, attach varchar(3) NOT NULL default 'yes', saycid varchar(3) NOT NULL default 'yes', hidefromdir varchar(3) NOT NULL default 'no', PRIMARY KEY (uniqueid), KEY mailbox_context (mailbox,context) ) TYPE=MyISAM; am i missing something? -- __ Edwin Lam [EMAIL PROTECTED] __ __ Systems Engineer, Office General, Inc. __ Ph: +1 415 439 4988 Fax: +1 415 283 3370 __ __ http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0xEF11A895 __ ___ --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] problems with a pri (E1)
Hi all, Our asterisk PBX, randomly restarts all the channels of the E1 connection. It sends this message There is no D-Channel, using channel 16 anyway.Then the asteisk recive (or it thinks it recives) yellow alarms at all the B-channels, after that it restart all the channels. When restarting the B-channels it cut all the conversations that is handling at that moment. Does anyone have an idea for what it is happening? We run a asterisk 1.2.1 on a HP Proliant ML310, PIV 3Ghz 2 Mb L2 cache, 1 Gb Ram. We have a TE210P digium card configured for E1. This pbx has been running for almost a moth before giving this problems, we have called our telco and seens that in their side all is ok. (Our telco is ONO- Spain) This is the zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow bchan = 1-15,17-31 dchan = 16 # Global data loadzone= es defaultzone = es this is the zapata.conf [trunkgroups] [channels] language=es context=from-pstn switchtype=euroisdn pridialplan=unknown signalling=pri_cpe busydetect=yes callprogress=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no channel=1-15,17-31 callerid=asreceived thanks in advance, Xavier Gil Estarellas. __ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.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
[Asterisk-Users] Exited non-zero
I am working on this app to dial two external numbers the second after the
first hangs up. I have simplified things down to:
exten = 3852,1,Dial(zap/g1/3964,10,g)
exten = 3852,2,Wait(2)
exten = 3852,3,Dial(zap/g1/7757,10,g)
exten = 3852,4,Hangup
Here is the debug:
-- Accepting call from '00' to '3852' on channel 0/23, span 1
-- Executing Dial(Zap/23-1, zap/g1/3964|10|g) in new stack
-- Called g1/3964
-- Zap/1-1 is ringing
-- Zap/1-1 answered Zap/23-1
-- Attempting native bridge of Zap/23-1 and Zap/1-1
Everything is OK here when someone picks up x3964. Then when x3964 is hung
up, the first Dial command executes fine, returning -1 (as per the docs
since it is disconnected by the far end). This causes a debug message of:
-- Hungup 'Zap/1-1'
== Spawn extension (from-internal, 3852, 1) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
And execution halts.
I need to figure out either how to get the dial command to return 0 or get
the system to continue execution of the script despite the non-zero return.
Has anyone dealt with this before?
Thanks in advance,
Wes
-Original Message-
From: Schochet, Wes
Sent: Friday, January 13, 2006 2:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Bridging app
OK - this is great! However, I'm showing my lack of depth / newness here.
Calls from internal SIP phones work perfectly. Calls from external sources
(my PBX) fail. Obviously, I have a dialplan / context problem, but I'd
appreciate a brief explanation and some direction from the group!
In extensions.conf, I have [from-pstn]. Under that section, I have included
[ext-postcall]. Then I have the following in an included file:
[ext-postcall]
exten = 3852,1,Answer
exten = 3852,2,Dial(zap/g1/8030,10,g)
exten = 3852,3,wait(5)
exten = 3852,4,Dial(zap/g1/8041,10,g)
exten = 3852,5,wait(5)
exten = 3852,6,NoOp(${DIALSTATUS})
exten = 3852,7,Hangup
The trunk group g1 is a T1 to my PBX and in the zapata.conf file I have the
entry
context = from-pstn
..
..
Group = g1
Here is the trace from both an internal extension (205) and an external
extension.
From SIP/205 ( Context=from-internal in sip.conf ) asterisk*CLI
-- Executing Answer(SIP/205-1d7b, ) in new stack
-- Executing Dial(SIP/205-1d7b, zap/g1/8030|10|g) in new stack
-- Called g1/8030
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/205-1d7b
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing Dial(SIP/205-1d7b, zap/g1/8041|10|g) in new stack
-- Called g1/8041
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/205-1d7b
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing NoOp(SIP/205-1d7b, ANSWER) in new stack
-- Executing Hangup(SIP/205-1d7b, ) in new stack
== Spawn extension (from-internal, 3852, 7) exited non-zero on
'SIP/205-1d7b'
-- Executing Macro(SIP/205-1d7b, hangupcall) in new stack
-- Executing ResetCDR(SIP/205-1d7b, w) in new stack
-- Executing NoCDR(SIP/205-1d7b, ) in new stack
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing Hangup(SIP/205-1d7b, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/205-1d7b' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/205-1d7b'
asterisk*CLI
From External coming in Zap/g1 (from-pstn) :
asterisk*CLI
-- Executing Answer(Zap/23-1, ) in new stack
-- Accepting call from '00' to '3852' on channel 0/23, span 1
-- Executing Dial(Zap/23-1, zap/g1/8030|10|g) in new stack
-- Called g1/8030
-- Zap/1-1 is ringing
-- Zap/1-1 answered Zap/23-1
-- Attempting native bridge of Zap/23-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (from-pstn, 3852, 2) exited non-zero on 'Zap/23-1'
-- Executing Macro(Zap/23-1, hangupcall) in new stack
-- Executing ResetCDR(Zap/23-1, w) in new stack
-- Executing NoCDR(Zap/23-1, ) in new stack
-- Executing Wait(Zap/23-1, 5) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'Zap/23-1'
in macro 'hangupcall'
== Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
asterisk*CLI
-Original Message-
From: trixter aka Bret McDanel [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 12, 2006 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bridging app
There might be a simplier way. a channel variable that holds the users
response, and a gotoif. You should be able to pass 'g' to dial which
according to http://www.voip-info.org/wiki-Asterisk+cmd+Dial
g: When the called party hangs up, exit to execute more commands in the
current context.
So the agent just hangs up and the IVR will continue with the caller into
your survey if they
[Asterisk-Users] test
test ___ --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 down because of cdr
Hello, After 2 weeks and 4 days without a problem, Asterisk went down. What happened is that I am using Asterisk 1.2.1 on a machine and have a MySQL for CDR on another machine. The machine with MySQL went down and the Asterisk box was unable to connect to MySQL. This made Asterisk to go down and it was unable to restart until MySQL was back. I know that Asterisk displays a lot of warnings, but still works, when the cdr table is corrupt. But isn't it a strange behaviour to go down when MySQL is down? Thank you Dov ___ --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] Max Number of #include statements
What's the maximum number of #include statments I can have in extensions.conf? I'm getting an error at the 11th one. I tried breaking twelve #include's into 2 different contexts, and still got the same error. These aren't nested includes... they're only one level deep. Thanks, Doug. ___ --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] Pre-made E1 crossver cables for the UK
Hi, just a note to let people know that I had NetShop make me some E1 crossover cables to replace my own dodgy crimpings, and they work perfectly =) The 3 metre version is £5.58 and their order code is CS000111/3 (I guess the /3 refers to the length..). They're at www.netshop.co.uk - 01753 691661. Cheers, Gavin. ___ --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] Distinctive ring?
Has anyone found a solution to this? On Sun, 27 Nov 2005 01:46 am, Kristof Hardy wrote: Kerry Garrison wrote: pain to configure) have 4 ring types. I am guessing that I would need to figure out how to tell this particular phone to use a different ring tone unless there is a way to send a stutter type ring to the phones. Hi Kerry, I'm also using grandstreams on a few places, have the 'same' issue/question. Afaik it can't be done with the current Grandsteam firmware. (at least, you can't command the phone to use tone X, like you can do with Cisco's) You can use the phone's built-in Distinctive Ring Tone: setting (Advanced settings), but I'm not aware of any 'wildcard' you can fill in there, I only got it to work when filling in an 'exact' number. It could be that the next firmware (should have arrived end of oct) gives us distinctive ring tones and working hint leds.. Let's hope.. If you do find a way to get any working, please report back to the list, meanwhile, i'm eagerly waiting for the firmware :) 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 -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --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] MeetMe greeting message.
Hi, all. My president wants to have a custom greeting for our bridge. So, I had it recorded (as foo.gsm), modified app_meetme.c to reflect the new filename, compiled, installed... and now get Jan 16 12:53:05 WARNING[14859] file.c: File foo does not exist in any format Jan 16 12:53:05 WARNING[14859] file.c: Unable to open foo (format ulaw): No such file or directory It's in the same directory (/usr/share/asterisk/sounds/) as the other greeting, with the same permissions and ownership. Is there something I'm missing? Thanks, -Ken ___ --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] asterisk1.2.1/PRI-E1 outbound call issues
Hello List,I have a following setup:1-Intel Zeon 3.0 Ghz dual Zeon capable board2-Ram 1GB3-OS SLES9 SP24-Asterisk-1.2.15-Wildcard TE110P(Using as E1) 6-Wildcard TDM03BPRI/E1 is up and running perfectly inbound /outbound calls goes perfectly in start but after sometime almost all outbound calls disconnected/hangup automatically for example user is dialout and he/she is talking then after sometime call disconnected inbound calls are also facing same issue but rate is very low sometimes while outbound calls becomes nightmare any idea here is my config:zaptel.conf:span=1,0,0,ccs,hdb3,crc4,yellowbchan=1-15,17-31dchan=16#dchan=31fxsks=33-35loadzone = usdefaultzone=us zapata.conf:[channels];calleridasreceived=yesusecallerid=yeshidecallerid=no;callwaitingcallerid=yes;threewaycalling=yes transfer=yescallwaiting=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesechotraining=800relaxdtmf=yes;faxdetect=incomingimmediate=no;Callgroup=1;Pickupgroup=1; ;context=from-pstn;switchtype=nationalsignalling=pri_cpe;faxdetect=incomingusecallerid=yescidsignalling=bellcidstart=ringpridialplan=nationalprilocaldialplan=nationalnationalprefix=0 localprefix=021busydetect=yesechocancel=yescallerid=yesechocancelwhenbridged=yesechotraining=800group=1channel=1-15,17-31;Callgroup=1;Pickupgroup=1context=incoming-analog group=2signalling=fxs_kschannel=33-35 ___ --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] Detecting Long PDD
Original Message From: Jean-Michel Hiver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 15, 2006 1:16 PM Subject: [Asterisk-Users] Detecting Long PDD Hi List, I've had some issues with some VoIP providers where either: 1 - There is massive PDD but finally the call goes through 2 - There is massive PDD but the call gets rejected anyways You might start by defining PDD. Most google hits for PDD is about autism... Leif ___ --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] loading zaptel drivers automatically upon reboot
On Fri, 2006-01-13 at 19:50 -0500, hugolivude wrote: Just installed Asterisk 1.2 on a brand new clean machine running RedHat 9.0. I have a TDM400 card inside. When I boot, the card seems dead. When I do: modprobe wctdm modprobe Zaptel the lights come on and all seems fine, until I reboot that is... After a reboot I have to repeat the modprobe. I shouldn't have to do a modprobe every re-boot should I? How do you get the drivers to load automatically? I've looked everywhere! - I tried running ztcfg but it did nothing, - I read a posting that spoke of editing rc.modules file, but I don't seem to have that file, - I tried removing everything that corresponds to zaptel, (including, but not limited to 'ztcfg', 'tor2' and 'tormenta' devices) from /etc/modules.conf. Again no luck Any ideas? For my TDM400 cards, I need to run the following commands: modprobe wcfxo modprobe wcfxs udevstart sleep 5 ___ --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] Advice Of Charge (AOC) ?
On Sun, 2006-01-15 at 10:32 +0100, Armin Schindler wrote: On Sun, 15 Jan 2006, Pisac wrote: Do Asterisk support Advice Of Charge (AOC) on ISDN lines? Do any ISDN drivers (bristuff, capi, vISDN, mISDN) support AOC? It is not implemented in chan_capi yet, but this is very easy. The question is what should be done with the AOC information? Just set some variable? As far as I know Asterisk has no API/structure for that. Hi Armin, Given the rather lackluster reaction to AOC so far I guess sticking it in a variable is the way to go. I don't know if KPN supports it. If they do and you decide to implement it then I'll be happy to test it for you. Regards, Patrick ___ --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] Set(LANGUAGE()=language throwing warnings
Hi, I tried setting a particular language through the agi framework using the following command EXEC Set(LANGUAGE()=%s)\n, language) // language contains EN or en On the asterisk command line I get the following warning: Jan 16 22:55:23 WARNING[4212]: res_agi.c:1085 handle_exec: Could not find application (Set(LANGUAGE()=EN)) strangely all IVR's are picked from the /var/lib/asterisk/sounds/en folder. I dont understand why asterisk throws this error. Does anyone have any idea whats going wrong? Regards, Danish ps: I am using asterisk 1.2.1 release ___ --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] CVS HEAD chanisavail not working for sip channel?
Dear all,
I have encountered problem with app_chanisavail for sip
channels. I have setup call-limit=1 in sip.conf as instructed, but when
making call to app_chanisavail, the channels did not increment correctly.
I end up dialing multiple times to the first channel only.
I think the ast_device_state(trychan) did not returned correctly.
Any idea?
Extensions.conf :
exten =
_1234.,1,ChanIsAvail(SIP/1SIP/2SIP/3SIP/4SIP/5|s)
exten = _1234.,n,Dial(${AVAILORIGCHAN}/0${EXTEN:4}|45)
exten = _1234.,n+101,busy
sip.conf :
[1]
type=friend
context=default
host=xxx.xxx.xxx.xxx
username=abcd
secret=abcd
port=5060
call-limit=1
fromuser=abcd
fromdomain=xxx.xxx.xxx.xxx
nat=yes
canreinvite=no
insecure=yes
insecure=very
disallow=all
allow=g723
allow=g729
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[Asterisk-Users] Starting from scratch
I have been tasked with moving our office from our junky old Nortel Meridian system to Asterisk. I will be keeping the T-1 PRI Voice circuits for the immediate future. My current intent is to purchase a nice Dell server to run everything, with a Digium TE110P PRI card. I also intend to run some version of Centos as the operating system. I have a mix of Centos 3 and Centos 4 boxes here and would like to keep that consistent. There will be about 25 extensions to start and up to another 10 in the next year. More than that and we need more office space. I already have ethernet running to all of the locations where phones will be, but in most cases only 1 port, which is already being used by a PC, so I will likely need a phone with 2 ports so I can daisy-chain off of it. Customization will be likely as we are a technology-heavy company and would like to be able to link incoming phone numbers to orders and comments in the database for the sales and customer service reps eventually. We have a programming department (I am sysadmin) and will be able to write the code to do this wither on the phone or on the rep's screen (pushed based on static IPs). I would like to keep the phones under $300 apiece (well under if possible). Questions: (1) Any advantage of Centos 3 or 4? (2) What phones would be best to get? (3) Any recommendation on a Dell server? I was thinking a PE1850 because of the dual power supplies and hardware RAID in a 1U chassis. (4) If I get outside sales agents working from home, what would be a good phone for them to get to hook into our system as a local extension? Thanks a bunch! Warren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phones unbeatable echo
Would you mind sharing with the list the tellabs hardware and how you got it up and running (ie pinouts etc)? On 1/15/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello Dan,I was fighting with echo on a number of circumstances, and came to thefollowing conclusions.If you are on a distant loop, or analog lines with issues, those issuesneed to be addressed or you need a workaround. In a few cases, I converted to ISDN-BRI, which has been one of my bestdecisions, because I get excellent quality as well as high-speed callcompletion...In one case, I put in an ADIT 600 channel bank, and still had serious echo problems. I tried and tried, but found no simple solution bymessing with the zapata drivers.Installing a hardware tellabs echocanceller totally solved the echo issue.I have the zapata.conf echocancellation totally off, and the lines sound great.These are also lines that are odd, meaning about 15K feet from the CO, with periodicinstabilities during rain/snow.I went through the various tweaks, milliwatt tests, etc, but only thehardware could solve it (and in minutes after installation as opposed to the hours I spend working with software).Depending on the amount of channels you have, you may consider achannelbank with tellabs, or one of the new digium analog cards with ec,though I have not used the new digiums yet myself.They are expensive solutions, but the best solutions too.I wish there were 4 port card that had great EC, but there isn't.Iwait for the day that we have pci-express voip cards at our disposal,that would be something...Asterisk would take off entirely at that point, since the latencies that cause so many problems would be gone,and the capacities would be so much higher.Just in case I went over your head here, sipsip should produce no echo.If it does there are other issues.If you are going analoganalog and hear no echo, I would have a look at the network itself.Regards,Greg-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dan ElderSent: Thursday, January 12, 2006 2:53 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP phones unbeatable echoHey all again, I'm wrestling with echo problems on our sip extensions.I've set these items in zapata.conf but tweaking these values doesn't seem to make much differenceechocancel=yesechocancelwhenbridged=yesechotraining=2500rxgain=8.0txgain=1.0are there other settings that can help me tame this beast? Beensearching but not turning up anything that'll work here. Thanks in advance.___--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 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] auto fallthrough hangup on 1.2.1
On Sunday 15 January 2006 00:07, Pisac wrote: I isolated problem, but I cannot find a cause. I think this is a bug! So, there is very very simplified dialplan which working in 1.0.9 but in 1.2.1 have that unexpected hangup: ;- exten = s,1,answer exten = s,2,digittimeout(0) exten = s,3,responsetimeout(15) exten = s,4,background(ivr-announce) exten = 1,1,goto(option1,s,1) exten = 2,1,goto(option2,s,1) exten = t,1,hangup exten = i,1,goto(s,1) ;- Did you read the file UPGRADE.txt in the 1.2.1 source treee ? There is a description of a global option that affects how * falls thru in the dialplan ? autofallthrough Paul When I press any digit DURING PLAYING ivr-announce (not after announce is finished), my line hangup: == Auto fallthrough, channel 'IAX2/someusername-2' status is 'UNKNOWN' -- Hungup 'IAX2/someusername-2' Where 'IAX2/...' is channel through I connected to IVR. If I connect through ZAP (ISDN/PSTN) then there is written 'ZAP/...', and if I get to this IVR context after some dialing of busy number, then ...status is 'BUSY' instead 'UNKNOWN'. If i change digit timeout to 1 sec: exten = s,2,digittimeout(1) then everything working as it should !!! *** So, conclusion is that problem with unexpected line hangup occuring only when digittimeout=0 and some DTMF digit is pressed during playing some voice file. IS THIS A BUG? *** My temporary solution is to set digittimeout=1. Any comment about this issue? Cheers. - Original Message - From: Pisac [EMAIL PROTECTED] I upgraded from 1.0.9 to 1.2.1 My IVR which worked perfectly on 1.0.9, now hangup with no reason (at least I could not find a cause) When this hangup happen, I can read: == Auto fallthrough, channel 'IAX/user-20' status is 'BUSY' This happening also with ZAP channels I'm really disappointed with 1.2.1, what is benefit from upgrade if I must spend couple days to get my system to work as it worked previously before upgrade (I think it should be named troublegrade). ___ --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 -- Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za Tel: +27 21 852 8812 Cel: +27 84 420 9282 Fax: +27 86 672 0563 -- ___ --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] New RPM packages for CentOS4.0
Greetings list, It's been a while since I've been able to focus on asterisk packaging but this weekend I took some time to audit and recompile packages for CentOS 4.2. You can find them here. ftp://ftp.linuxsys.com/ftp/pub/releases/CentOS-4.0 You have your choice of 1.2.1 or 1.0.10 releases. If you need zaptel modules then install this kernel as well: ftp://ftp.linuxsys.com/ftp/pub/releases/CentOS-4.0/kernel SRPMS are available for those wishing to recompile zaptel against their own kernel. Features of this release - 1.2.1 patched with spandsp-0.0.2pre22 - 1.0.10 patched with spandsp-0.0.2pre21 - init script launches safe_asterisk by default - compiled to include cdr_addon_mysql.so and format_mp3.so - asterisk console is automatically launched on pseudo tty8 - zaptel init script configs are moved to /etc/sysconfig/zaptel - tested to work with AMP (required software available as rpms) Other packages released astcc-40-1.RHEL4.LSE.i386.rpm asterisk-sounds-31-1.RHEL4.LSE.i386.rpm gtkiaxyprov-17-1.RHEL4.LSE.i386.rpm gastman-54-1.RHEL4.LSE.i386.rpm iaxyprov-15-1.RHEL4.LSE.i386.rpm lame-3.96.1-RHEL4.LSE.1.i386.rpm lame-devel-3.96.1-RHEL4.LSE.1.i386.rpm perl-DBD-mysql-3.0002-1.RHEL4.LSE.i386.rpm perl-IPC-Signal-1.00-1.RHEL4.LSE.i386.rpm perl-mime-construct-1.9-1.RHEL4.LSE.i386.rpm perl-Net-Telnet-3.03-1.RHEL4.LSE.i386.rpm perl-Proc-WaitStat-1.00-1.RHEL4.LSE.i386.rpm As you should expect theses packages come with no warranty whatsoever but I would like some feedback so please feel free to contact me via email - amcroryaTlinuxsysDotcom. Best Regards, Andrew McRory - President / CTO Linux Systems Engineers, Inc. Located in beautiful Tallahassee, Florida (850) 224-5737 x2005 ___ --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] Reducing echo on FXS port
TryIn chan_zap.c change the following line:#define READ_SIZE 160to#define READ_SIZE 16In zapata.confjitterbuffers=40This will also increase system load by a factor of 10. 2006/1/15, Aryanto Rachmad [EMAIL PROTECTED]: Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 -- FXS (TDM400P) -- Asterisk -- SIP GW -- PSTN -- Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echoto maximum with the following settings for my FXS port on zapata.conf: rxgain=-8.0txgain=2.0echocancel=256echotraining=500 Butit isstill not entirely eliminated as westill sometimes hear the last syllables, with the level ofmaybe 5% of the original sound. What I did was just playing around with the values of those parameters, use ztmonitor to have the FXS rx/tx signalvisualised and use only my ears to check it. I think my ears are fine :), as I dothis because my friends complain about the echo they hear. Does anybody know a better method tofind the best value forthose parameters? There is no echo on phone2 when Iusesoftphone like this: PC(X-Lite) -- Asterisk -- SIP GW -- PSTN -- Phone2 The following is the version of asterisk I am using: CLI show version Asterisk SVN-branch-1.2-r7999 built by root @ atvie-asterisk on a i686 running Linux on 2006-01-13 06:15:02 UTC And I set the echo canceller in zconfig.h to ECHO_CAN_MG2. Cheers, Anto ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pickup Button
Our current telephone system supports a function called pickup, where a user can press the pickup button dial an extension and take a call that was meant for that extension. (a teacher calls the principal however he /she is not in so the secretary presses the pickup button dials his extension and it routes the call to her phone) How can i do this with programming? Also we will be using all Cisco 7940 phones, their is an extra button on the display can that be set to be a function? Such as pickup? Thank You Johnathan Falk Network Administrator Clinton Community Schools ___ --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] Echo
On Sun, 2006-01-15 at 22:23 -0500, Steve Totaro wrote: Just checking Just checking :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] Support for RFC3323?
Does anybody know, if asterisk support the rfc 3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)?? I'm working with a Softswitch wich works with this rfc, and I don't know jet how to dissable this functionality. This is a problem becouse the SS do not pass the ANI in the interworking SIP-SS7 (only in this direcction). Regards Miguel ___ --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] PHPAGI daemon/background task?
Hum, i still have not a clear idea about where it gets stuck. Lets see. 1. Once a minut, check to see if any meetme conferences are active. This is done by connecting each minute to the manager, or it stays connected between each minute? 2. It registers an event handler for MeetMeLeave it waits until the event arrives then? how do you wait? are you using socket_select() call to prevent you from calling socket_read() when nothing to read? regards On 1/14/06, Dan Austin [EMAIL PROTECTED] wrote: The script has two functions- 1. Once a minute check to see if any MeetMe conferences are active and list the participants of any active conferences. 2. It registers an event_handler for MeetMeLeave and processes the output. The script simply loops issues manager commands. If command fails, it exits the inner loop, sleeps for awhile and tries to reconnect to the manager. If it fails, it sleeps more and repeats the process. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Friday, January 13, 2006 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PHPAGI daemon/background task? im not familiar with phpagi-asmanager.php, but i guess thats because asterisk has nothing to send you and you are using blocking mode sockets, hence socket_read() is blocking. phpagi-asmanager.php give us more pointings about what you want to do and we will be able to help you. Currently im developing a php daemon to listen events and work as proxy for other clients that do real work depending on the events, something like that you want to do? On 1/13/06, Dan Austin [EMAIL PROTECTED] wrote: I have a script that I want to leave running in the background to handle specific manager events. I'm running into a problem where it gets stuck in the wait_response function in phpagi-asmanager.php and the PHP maximum execute timeout kills the script. The script doesn't interact with the dialplan, so I cannot launch it from within Asterisk. Any pointers would be appreciated. I did look through the wiki and gave google a chance, but the results found didn't really suggest a solution. Dan ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamic conference - add participants
Hello all, I need to create dynamic conferences with variable number of patricipants. My users use custom SIP softphone and I want to implement fast conference creation/moderation. I search through voip-info.org examples and found some useful information in this page: http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro But this solution uses ugly way to add new participants. Any ideas how can I add several participants at a heat? I think this may be done though Asterisk manager interface, but I think there must be a way to achieve utilizing Dialplan features. ___ --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] Random Disconnects
On 1/14/06, C F [EMAIL PROTECTED] wrote: I'm using Sipura 3000 as well, however I will have to wait until Monday about the Switch I'm not sure. So far it looks like Sipura is at fault. In the mean time I would like to hear from others using the Sipura 3000 FXO if they have the same problem. For now I am experimenting with allowing reinvites between the SPA-3000 FXO port and a couple of other extensions. I sent several inquiries to people who complained of this problem in the past (6+ months ago). I haven't heard back from them. 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
Re: [Asterisk-Users] SIP RTP
It just re-directs the RTP stream. The SIP stream still goes through *. Mike Hammett wrote: According to this page: http://www.asterisk.org/doxygen/Config_sip.html canreinvite=yes redirects just the RTP. I was under the impression that the entire SIP connection got redirected, therefore losing accounting ability. Could someone clarify this? --Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Network stuff you didn't know http://www.networkoblivion.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] dnid
Hi all! I'm in the process of configuring an Asterisk server here that, based on which number was called, should send calls to different extensions: 913 - 1 - ext. 1 913 - 2 - ext. 2 913-1 913-2 being 2 (of the) numbers we have coming in to our system via our VoIP hosting provider. The config used here is based on Asterisk at home, so it includes also the dialparties.agi script. This script sees and identifies the correct dnid, but I am having some trouble to get the dialplan to act on this value. The info in the Wiki ( http://www.voip-info.org/tiki-index.php?page=DNID ) is not of much help either. Anyone here with any suggestions? Regards, Evert ___ --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: RE: RE: RE: Spandsp
I have included logger.conf and now I see that problem is with loading libspandsp.so. I dont have that file in /usr/src/asterisk-1.2.1/apps Can you tell me where do you have it? Does it means that spandsp wasn't installed corectly? This is what I get when I try to start * with logger.conf. [app_txfax.so]Jan 16 10:01:35 WARNING[7933]: loader.c:325 __load_resource: libs pandsp.so.0: cannot open shared object file: No such file or directory Jan 16 10:01:35 WARNING[7933]: loader.c:325 __load_resource: libspandsp.so.0: ca nnot open shared object file: No such file or directory Jan 16 10:01:35 WARNING[7933]: loader.c:554 load_modules: Loading module app_txf ax.so failed! Jan 16 10:01:35 WARNING[7933]: loader.c:554 load_modules: Loading module app_txf ax.so failed! -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] distorted native music on hold
Hello, Using asterisk-1.2.1 I am trying to convert my music-on-hold files from .wav to alaw: % sox moh.wav -r 8000 -c 1 moh.al resample -ql The file sounds fine when listened with: % sox mox.al -t ossdsp /dev/dsp But when listened through asterisk with an alaw SIP phone the sound is clicky and too fast. Did I forget something in my conversion command? -- [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] cmd Dial parameters
Hi, For the dial application, parameter g is described as g: When the called party hangs up, exit to execute more commands in the current context. I want the following priority (or at least a priority I can jump to) to be executed anyway, it doesn't matter which party hang up. Is there a way to do so? Thank you Dov ___ --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: Mediatrix windows-based setup?
Please stop replaying to mesage. If you plan to open thread do so by writing mail to this address asterisk-users@lists.digium.com -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Test to see if I'm still on list...
I was having problems too. Mine stopped at 5:19am MST this morning and just picked up a few minutes ago. Isn't the first time it's happened either. -Original Message- From: Francesco Peeters [mailto:[EMAIL PROTECTED] Sent: Monday, January 16, 2006 3:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Test to see if I'm still on list... As I haven't received any posts since yesterday... ___ --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] RJ21-RJ11
Ing. Germán González B. wrote: Hi!! I'm looking for an adapter RJ21 to 24 RJ11 for a TDM2400. Somebody can help me with some sugestions? Thks!!! --- Germán González http://leon.podernet.com.mx --- ___ --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 http://shop3.outpost.com/product/1729164?site=sr:SEARCH:MAIN_RSLT_PG -- Roman Volf Keystreams Internet Solutions [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] Re: RE: RE: RE: Spandsp
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Download hylafax, and iaxmodem. Set up a friend extension as iax, and let it rip... it's a slam dunk. I think I have found the real problem source (spandsp, not txfax) and maybe now I solve it. If I don't manage, I will surtnely lisen your suggest. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo
Steve Totaro wrote: Just checking ___ --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 I was asking myself the saim thing... -- Razvan Turtureanu NOC Engineer Mobil: 072.3637714 E-mail: [EMAIL PROTECTED] S.C. Edata S.R.L., Sucursala Bucuresti Str. Toamnei nr. 68, et. III, sector 2 Tel: +4031.401.6828 Fax: +4031.401.6829 E-mail: [EMAIL PROTECTED] Web: www.edata.ro This e-mail is confidential and may contain legally privileged information. If you are not the intended recipient, you should not copy, distribute, disclose or use the information it contains. Please e-mail the sender immediately and delete this message from your system. E-mails are susceptible to corruption, interception and unauthorised amendment; we do not accept liability for any such changes, or for their consequences. You should be aware, that a company may monitor your emails and their content. Acest mesaj este confidential si poate contine informatii protejate legal. Daca nu sinteti destinatarul intentionat, nu trebuie sa copiati, difuzati, dezvaluiti sau utilizati informatiile pe care acesta le contine. Va rugam sa retransmiteti imediat mesajul expeditorului si sa-l stergeti din sistemul dvs. Mesajele sint pasibile de denaturare, interceptie sau modificare neautorizata; nu ne asumam raspunderea pentru nici o asemenea eventuala schimbare sau pentru consecintele acesteia. Trebuie sa stiti ca o companie va poate monitoriza mesajele si continutul acestora. ___ --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] agi debug - unable to set normal priority
Hello! In my agi-debug i get the following error-message: AGI Rx Jan 16 14:45:52 WARNING[18299]: asterisk.c:788 ast_set_priority: Unable to set normal priority AGI Tx 510 Invalid or unknown command AGI Rx SET VARIABLE MODCLI 00434345452 the agi i call is a very simple shellscript that simply removes wrong charakters: #!/bin/bash modcli=`echo $1 | sed -e 's/#//g' -e 's/*//g'` #echo $modcli echo SET VARIABLE MODCLI $modcli the script works as expect, sending the modified variable back to asterisk... anyone knows what this error-message means? regards christian ___ --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
FW: [Asterisk-Users] confirmation
Tzafrir, Did you get my e-mail with the zaptel.conf Zapata.conf I want to confirm that? Thanks, Carlos Alperin ___ --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: 1.2.1 Silence suppression is disabled whatthehell?
On Sunday, January 15, 2006 12:21 AM Tony Mountifield wrote: In article [EMAIL PROTECTED], Pisac [EMAIL PROTECTED] wrote: I've found something here: http://bugs.digium.com/view.php?id=5374 but I don't understand how this can be connected to my problem :-( It looks like the maintainer of the BRIstuff distribution might have decided that patch was worth including, even though it is not in the standard 1.2.1. That does give scope for confusion though! Look at the CHANGES. I was the one who convinced kapjeod to put that patch in the current bristuff distribution. So yes: It is in bristuff as of 1F: 0.3.0-PRE-1f - THIS IS GETTING CLOSER TO A STABLE RELEASE, USE IN PRODUCTION AT YOUR OWN RISK! - merged patch for bug 5697 (meetme) - merged patch for bug 5374 (asynchronous generation of outgoing frames) - _finally_ fixed sending-nonRFCcompliant-SIP-NOTIFYs bug (asterisk, extension states) - some debug output clean ups in libpri Kind 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] echo tail stats
Hi Eric, If you have a Sangoma Card, you can find use the echo debugging tools that come with our wanpipe-beta1w-2.3.4 drivers or later (ftp://ftp.sangoma.com/linux/custom/2.3.4/wanpipe-beta1y-2.3.4.tgz). You will find instructions on how to use our echo debugging tools at: http://sangoma.editme.com/wanpipe-linux-asterisk-debugging David Yat Sin Sangoma Technologies (905) 474-1990 x119 (800) 388-2475 x119 Fax: (905) 474 9223 MSN: [EMAIL PROTECTED] Email: [EMAIL PROTECTED] Wiki: sangoma.editme.com From: Eric Bishop [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] echo tail stats Date: Sun, 15 Jan 2006 10:10:50 +1100 Does anyone know how to determine the echo tail size (in ms) of a particular call? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX voice distortion with full upload channel /SIP ok
On Samstag, 14. Januar 2006 1:47 tim panton wrote: That is weird, you would expect IAX to do better than SIP (bandwidth wise) My point exactly. 1) are you sure IAX trunking is actually happening ? It shows (T) in iax2 show so I am pretty sure. Timestamps are enabled as well. 2) what codecs are you using. Are the codecs the same for IAX as for sip? G.711 alaw and yes the same for IAX and SIP. 3) is it possible that some of the network hardware is 'sip aware' I strongly doubt it. Our firewall is but only regarding to opening the correct RTP ports for a SIP call. No traffic shaping is done on that end. 4) How many simultaneous calls are you running between the 2 endpoints? Happens with one call. 5) What happens if you turn trunking off ? No change. Kind 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
[Asterisk-Users] Unable to open master device '/dev/zap/ctl'
Hi, I'm running [EMAIL PROTECTED] with a TDM2400 When i try to load the wctd24xx always get the same error: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected I try to load the modules manualy: [EMAIL PROTECTED] ~]# modprobe zaptel [EMAIL PROTECTED] ~]# modprobe wctdm24xxp Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected /var/log/messages: Jan 16 07:53:30 asterisk1 kernel: Zapata Telephony Interface Registered on major 196 Jan 16 07:55:12 asterisk1 kernel: ACPI: PCI interrupt :02:0c.0[A] - GSI 20 (level, low) - IRQ 217 Jan 16 07:55:12 asterisk1 kernel: PCI Config reg is 02900117 Jan 16 07:55:12 asterisk1 kernel: WCTDM2400P: New Reg: fe59! Jan 16 07:55:12 asterisk1 kernel: Detected REG0: 0100 Jan 16 07:55:12 asterisk1 kernel: Detected REG1: 7849 Jan 16 07:55:12 asterisk1 kernel: Detected REG2: 001d Jan 16 07:55:12 asterisk1 kernel: (pre) Reg fc is 5027 Jan 16 07:55:12 asterisk1 kernel: (post) Reg fc is 5024 Jan 16 07:55:12 asterisk1 kernel: Detected REG2: Jan 16 07:55:12 asterisk1 kernel: wctdm2400p: reg is a04c0004 Jan 16 07:55:12 asterisk1 kernel: Resetting the modules... Jan 16 07:55:12 asterisk1 kernel: During Resetting the modules... Jan 16 07:55:12 asterisk1 kernel: After resetting the modules... Jan 16 07:55:13 asterisk1 kernel: Port 1: Not installed Jan 16 07:55:13 asterisk1 kernel: Port 2: Not installed Jan 16 07:55:13 asterisk1 kernel: Port 3: Not installed Jan 16 07:55:13 asterisk1 kernel: Port 4: Not installed Jan 16 07:55:13 asterisk1 kernel: Port 5: Not installed Jan 16 07:55:14 asterisk1 kernel: Port 6: Not installed Jan 16 07:55:14 asterisk1 kernel: Port 7: Not installed Jan 16 07:55:14 asterisk1 kernel: Port 8: Not installed Jan 16 07:55:14 asterisk1 kernel: Port 9: Not installed Jan 16 07:55:14 asterisk1 kernel: Port 10: Not installed Jan 16 07:55:15 asterisk1 kernel: Port 11: Not installed Jan 16 07:55:15 asterisk1 kernel: Port 12: Not installed Jan 16 07:55:15 asterisk1 kernel: Port 13: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:16 asterisk1 kernel: Port 14: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:16 asterisk1 kernel: Port 15: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:17 asterisk1 kernel: Port 16: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:17 asterisk1 kernel: Port 17: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:18 asterisk1 kernel: Port 18: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:18 asterisk1 kernel: Port 19: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:19 asterisk1 kernel: Port 20: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:19 asterisk1 kernel: Port 21: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:19 asterisk1 kernel: Port 22: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:20 asterisk1 kernel: Port 23: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:20 asterisk1 kernel: Port 24: Installed -- AUTO FXO (FCC mode) Jan 16 07:55:20 asterisk1 kernel: VPM: Chip 0: ver 33 Jan 16 07:55:20 asterisk1 kernel: VPM: U-law mode Jan 16 07:55:20 asterisk1 kernel: VPM: Chip 1: ver 33 Jan 16 07:55:20 asterisk1 kernel: VPM: Chip 2: ver 33 Jan 16 07:55:20 asterisk1 kernel: VPM: Chip 3: ver 33 Jan 16 07:55:21 asterisk1 kernel: VPM: DTMF threshold set to 1250 Jan 16 07:55:21 asterisk1 kernel: VPM: Present and operational Jan 16 07:55:21 asterisk1 kernel: Found a Wildcard TDM: Wildcard TDM2400P Prototype (24 modules) Seems that fails the ztcfg command in the modules installations. But i dont know how to solve it Any Ideas?? here is my zapata.conf [EMAIL PROTECTED] ~]# cat /etc/zaptel.conf fxsks=13-24 loadzone= us defaultzone = us Thanks in advance. -- Manuel Casal [EMAIL PROTECTED] [EMAIL PROTECTED] Sistemas de Información y Protección de Datos, S.L. Telf. + 34 902 678006 e-mail: [EMAIL PROTECTED] web: http://www.e-sistemas.net smime.p7s Description: S/MIME Cryptographic 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] IAX voice distortion with full upload channel /SIP ok
On Saturday, January 14, 2006 2:45 PM Rich Adamson wrote: The iax problems tend to be oriented around version issues. Many of the itsp's have added whatever functionality they needed to asterisk to support their operation, and upgrading their code to the latest levels is not a trevial task. Misunderstanding: I am talking about my private * against our company *. Both are running the exact same version of Asterisk. Given the changes that have occurred in the iax code over the last year or so, mismatches in iax versions are known to cause significant audio quality issues. Turning off the jitterbuffer, trunk=no, etc, is oftentimes the only way to get close to reasonable audio quality. I already tried this and this is not helping. 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
[Asterisk-Users] Asterisk for Call Center (missing reference)
Hi Folks, I've been searching for an specific feature on asterisk and I found an e-mail from John Todd asking for the same thing. http://lists.digium.com/pipermail/asterisk-users/2004-May/045882.html To be able to listen (zapbarge, zapscan or chanspy) an specific channel and can talk to one side (the operator). That feature is very usefull in call centers in Brazil so if you want to use Asterisk as a Call Center PBX you have to support it. John Todd post it in May 2004 so perhaps now (Jan 2006) it's possible or there is another app (commercial?) that can support it. John: have you found a solution for your question? if so, please let me know! Thanks in advance, -- Rodrigo P. Telles [EMAIL PROTECTED] IT Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group ___ --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 for Call Center
Hi Folks, I've been searching for an specific feature on asterisk and I found an e-mail from John Todd asking for the same thing. To be able to listen (zapbarge, zapscan or chanspy) an specific channel and can talk to one side (the operator). That feature is very usefull in call centers in Brazil so if you want to use Asterisk as a Call Center PBX you have to support it. John Todd post it in May 2004 so perhaps now (Jan 2006) it's possible or there is another app (commercial?) that can support it. John: have you found a solution for your question? if so, please let me know! Thanks in advance, -- Rodrigo P. Telles [EMAIL PROTECTED] IT Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group ___ --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] Reducing echo on FXS port
Aryanto Rachmad wrote: Hello everybody, I am sorry to bring this up again if this kind of echo issue has ever discussed. Phone2 in below call path experiences quite annoying echo: Phone1 -- FXS (TDM400P) -- Asterisk -- SIP GW -- PSTN -- Phone2 It is annoying as on phone2, we can hear the whole words we say with the level of maybe 25% of the original sound. I can reduce the echo to maximum with the following settings for my FXS port on zapata.conf: rxgain=-8.0 txgain=2.0 echocancel=256 echotraining=500 But it is still not entirely eliminated as we still sometimes hear the last syllables, with the level of maybe 5% of the original sound. What I did was just playing around with the values of those parameters, use ztmonitor to have the FXS rx/tx signal visualised and use only my ears to check it. I think my ears are fine :), as I do this because my friends complain about the echo they hear. Does anybody know a better method to find the best value for those parameters? There is no echo on phone2 when I use softphone like this: PC(X-Lite) -- Asterisk -- SIP GW -- PSTN -- Phone2 Are you sure that X-Lite is not running an echo can? I'd say it's more likely that the SIP GW is causing the echo and that when you use X-Lite, it's echo cans are removing the echo. Try to make a call from Phone-FXS-Asterisk-X-Lite I suspect there will be no echo. BTW what is the SIP Gateway? -- 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] strange voicemail issue
with this in extensions.conf: exten = xxx,x,voicemail([EMAIL PROTECTED]) I get this in the log: -- Executing VoiceMail(SIP/officeata1-5836, [EMAIL PROTECTED]) in new stack Jan 16 09:20:50 WARNING[2700]: app_voicemail.c:2379 leave_voicemail: No entry in voicemail config file for 'ales' no voicemail entry for ales? why is the first 's' chopped off? To make it more interesting, if I add the |s option thusly then everything works fine. exten = xxx,x,voicemail([EMAIL PROTECTED]|s) this is version 1.2.0 Anyone have any comments? ___ --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] Problem with calls starting from a legacy PBX
Hi, I have this setup: E1 PRI PSTN -- Asterisk -- Alcatel PBX - analog phones Can someone tell me what's wrong with this call initiating from an analog phone connected to Alcatel PBX? It dies with NOANSWER but all works if I call other destination numbers. Dialplan is a simple Dial(zap/g1/0984465691) statement. At the end you'll find also zapata.conf. # Protocol Discriminator: Q.931 (8) len=28 Call Ref: len= 2 (reference 19387/0x4BBB) (Originator) Message type: SETUP (5) [04 03 90 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 05 00 81 32 32 30] Calling Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation permitted, user number passed network screening (1) '220' ] [70 05 80 30 39 38 34] Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '0984' ] -- Making new call for cr 19387 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 30 (cs0, Progress Indicator) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 19387/0x4BBB) (Terminator) Message type: SETUP ACKNOWLEDGE (13) [18 03 a9 83 9d] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 29 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] -- Accepting overlap call from '220' to '0984' on channel 0/29, span 2 -- Starting simple switch on 'Zap/60-1' Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 19387/0x4BBB) (Originator) Message type: INFORMATION (123) [70 02 80 34] Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '4' ] -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 19387/0x4BBB) (Originator) Message type: INFORMATION (123) [70 02 80 36] Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '6' ] -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 19387/0x4BBB) (Originator) Message type: INFORMATION (123) [70 02 80 35] Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '5' ] -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 19387/0x4BBB) (Originator) Message type: INFORMATION (123) [70 02 80 36] Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '6' ] -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 19387/0x4BBB) (Originator) Message type: INFORMATION (123) [70 02 80 39] Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '9' ] -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 19387/0x4BBB) (Originator) Message type: INFORMATION (123) [70 02 80 31] Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '1' ] -- Processing IE 112 (cs0, Called Party Number) -- Executing NoOp(Zap/60-1, -- Thematica called from 0984899220 --) in new stack -- Executing Dial(Zap/60-1, Zap/g1/0984465691) in new stack -- Making new call for cr 33047 -- Requested transfer capability: 0x10 - 3K1AUDIO Protocol Discriminator: Q.931 (8) len=34 Call Ref: len= 2 (reference 279/0x117) (Originator) Message type: SETUP (5) [04 03 90 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 8d] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified
[Asterisk-Users] chan_capi-cm and DID
Hi all, i have asterisk 1.0.9 with an Eicon Diva 4bri and chan_capi-cm-0.6. I have 2 NTBAs (one with did and one without). When using the one without did, i am able to place outgoing and incoming calls. When i use the NTBAs with did i have a layer 2 error. Anyone an idea? -- Executing Dial(SIP/2004-9634, CAPI/g1/43XX) in new stack data = g1/43XX parsed dialstring: 'g1' '43XX' '' capi request group = 2 parsed dialstring: 'g1' '43XX' '' == EICON: Call CAPI/EICON/43XX-6 (pres=0x00, ton=0x00) CONNECT_REQ ID=001 #0x000c LEN=0065 Controller/PLCI/NCCI= 0x1 CIPValue= 0x10 CalledPartyNumber = 8043XX CallingPartyNumber = 00 80 22EyeBeam22 3c20043e CalledPartySubaddress = default CallingPartySubaddress = default BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default BC = default LLC = default HLC = default AdditionalInfo BChannelinformation= 00 00 Keypadfacility = default Useruserdata = default Facilitydataarray = default -- Called g1/43XX CONNECT_CONF ID=001 #0x000c LEN=0014 Controller/PLCI/NCCI= 0x201 Info= 0x0 -- EICON: received CONNECT_CONF PLCI = 0x201 DISCONNECT_IND ID=001 #0x0011 LEN=0014 Controller/PLCI/NCCI= 0x201 Reason = 0x3302 DISCONNECT_RESP ID=001 #0x0011 LEN=0012 Controller/PLCI/NCCI= 0x201 CAPI INFO 0x3302: Protocol error layer 2 == EICON: CAPI Hangingup == EICON: Interface cleanup PLCI=0x201 == No one is available to answer at this time my capi.conf looks like: [DID] controller=1,2,3,4 isdnmode=did incomingmsn=* softdtmf=on relaxdtmf=on accountcode= context=DID echocancel=yes ;echocancelold=yes devices=2 group=1 __ 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
[Asterisk-Users] RX/TXgain on bristuff/zaptel ?
Do bristuffed zaptel (zaphfc) supporting rxgain/txgain in zapata.conf? I'm changing rxgain in zapata.conf, and reloading zaptel, but sound level on ISDN(HFC) is always the same (loud). ___ --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] uplink call quality issues
Hi Can someone please help with the following, We are using [EMAIL PROTECTED] 1.5 and SIP trunks to communicate to the PSTN network. We are having some problems with the call quality. Although we can hear the other person's voice quite clear when making or receiving a call, we get complaints from the people on the other end saying that our voices sound very unclear, low and that the voice drops, therefore people on the other end can not understand what we are saying. But as I said in our end their voices sound clear. I have checked network wise and found no latency problems within our small LAN, with our VoIP provider and reaching their SIP server's IP address, also the CPU load in the asterisk server has been graphed and does not exceed the normal CPU load levels Any assistance will be very much appreciated PolAus _ Buy now @ Tradingpost.com.au http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fad%2Eau%2Edoubleclick%2Enet%2Fclk%3B24875379%3B12369854%3Ba%3Fhttp%3A%2F%2Fwww%2Etradingpost%2Ecom%2Eau%3Freferrer%3DnmsnHMetagv1_t=752643439_r=hotmailtagline_m=EXT ___ --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] automon - one touch record
Hello Kevin, Thank you for your response. I commented out DYNAMIC_FEATURES and moved the 'Ww' option to the Dial() instead of Queue() and now it works. Regards Jenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Friday, January 13, 2006 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] automon - one touch record Jennifer Hales wrote: I am unable to get automon recording to work; can someone advise me what I am doing wrong? When I do *1 all I see in the CLI screen is attempting native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call record generated in /var/spool/asterisk/monitor/. I can't explain why it's not working, but DYNAMIC_FEATURES is not necessary if you are providing the 'wW' options to the Queue application as you are. Can you try this with a regular Dial() call instead, to eliminate the queue application? ___ --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] ztdummy inaccuracy on linux-2.6
Hello, I have some ugly numbers given by zttest for ztdummy on an AMD64 box running linux-2.6.15 compiled for Athlon64. linux-2.6.15, zaptel/branches/1.2 r900, jiffies ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.987793% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.987793% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.987793% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% --- Results after 136 passes --- Best: 99.987793 -- Worst: 99.975586 -- Average: 99.975853 linux-2.6.15, zaptel/branches/1.2 r900, RTC Opened pseudo zap interface, measuring accuracy... 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.951172% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.951172% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% 99.938965% 99.963379% --- Results after 96 passes --- Best: 99.963379 -- Worst: 99.938965 -- Average: 99.952942 linux-2.6.15, zaptel/branches/1.2 r900+patch bugs.digium.com/view.php?id=5971, RTC Opened pseudo zap interface, measuring accuracy... 99.987793% 99.719238% 99.707031% 100.00% 99.890137% 99.865723% 100.00% 99.987793% 100.00% 99.975586% 100.00% 99.987793% 99.975586% 100.00% 100.00% 99.987793% 99.975586% 99.768066% 99.768066% 99.987793% 99.926758% 99.926758% 99.987793% 99.975586% 100.00% 99.975586% 99.987793% 100.00% 100.00% 99.975586% 99.938965% 100.00% 99.975586% 99.816895% 99.816895% 100.00% 99.987793% 99.975586% 100.00% 99.975586% 100.00% 99.987793% 100.00% 99.975586% 99.987793% 100.00% 99.719238% 99.707031% 99.987793% 99.877930% 99.865723% 100.00% 99.987793% 99.975586% 100.00% 99.987793% 100.00% 99.975586% 100.00% 99.987793% 99.987793% 100.00% 99.768066% 99.768066% 99.975586% 99.938965% 99.926758% 99.975586% 100.00% 99.987793% 100.00% 99.975586% 100.00% 99.975586% 99.987793% 100.00% 100.00% 99.987793% 99.829102% 99.816895% 99.975586% 99.987793% 99.975586% 100.00% 99.987793% 99.975586% 100.00% 99.975586% 99.987793% 100.00% 100.00% 99.719238% 99.694824% 100.00% 99.890137% 99.877930% 99.987793% 100.00% 99.975586% 99.987793% 100.00% 99.975586% 99.987793% 99.987793% 100.00% 99.975586% 100.00% 99.780273% 99.755859% 100.00% 99.938965% 99.938965% 99.975586% 100.00% 99.987793% 99.975586% 100.00% 99.987793% 100.00% 99.975586% 100.00% 99.987793% 99.975586% 99.816895% 99.816895% 100.00% 100.00% 99.975586% 99.975586% 99.987793% 100.00% 100.00% 99.987793% 99.975586% 100.00% 99.975586% --- Results after 136 passes --- Best: 100.00 -- Worst: 99.694824 -- Average: 99.951973 HW: Tyan Tomcat K8E, Athlon64 3000+, 1GB RAM, 3ware 8006, 2x Maxtor HDD SW: Ubuntu 5.10, linux-2.6.15, zaptel from 1.2 branch Any idea what can be wrong? Thanks in
[Asterisk-Users] List
The list is very quiet today - almost too quiet PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phones unbeatable echo
I have an install with the Digium TDM2400 with the EC module and even though Digium techs have spent well over 10 hours tweaking and tweaking the call quality is so bad we are ready to chuck it. I think that you were on the right track below in implieing that a different solution may be required at different locations based on the quality and performance of the phone lines. That has certainly been my experience so far. 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 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 15, 2006 12:27 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] SIP phones unbeatable echo Hello Dan, I was fighting with echo on a number of circumstances, and came to the following conclusions. If you are on a distant loop, or analog lines with issues, those issues need to be addressed or you need a workaround. In a few cases, I converted to ISDN-BRI, which has been one of my best decisions, because I get excellent quality as well as high-speed call completion... In one case, I put in an ADIT 600 channel bank, and still had serious echo problems. I tried and tried, but found no simple solution by messing with the zapata drivers. Installing a hardware tellabs echo canceller totally solved the echo issue. I have the zapata.conf echo cancellation totally off, and the lines sound great. These are also lines that are odd, meaning about 15K feet from the CO, with periodic instabilities during rain/snow. I went through the various tweaks, milliwatt tests, etc, but only the hardware could solve it (and in minutes after installation as opposed to the hours I spend working with software). Depending on the amount of channels you have, you may consider a channelbank with tellabs, or one of the new digium analog cards with ec, though I have not used the new digiums yet myself. They are expensive solutions, but the best solutions too. I wish there were 4 port card that had great EC, but there isn't. I wait for the day that we have pci-express voip cards at our disposal, that would be something... Asterisk would take off entirely at that point, since the latencies that cause so many problems would be gone, and the capacities would be so much higher. Just in case I went over your head here, sipsip should produce no echo. If it does there are other issues. If you are going analoganalog and hear no echo, I would have a look at the network itself. Regards, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Elder Sent: Thursday, January 12, 2006 2:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP phones unbeatable echo Hey all again, I'm wrestling with echo problems on our sip extensions. I've set these items in zapata.conf but tweaking these values doesn't seem to make much difference echocancel=yes echocancelwhenbridged=yes echotraining=2500 rxgain=8.0 txgain=1.0 are there other settings that can help me tame this beast? Been searching but not turning up anything that'll work here. Thanks in advance. ___ --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] automated response
Thank you for your email. Currently I am on vacation from 1/16/2006 to 1/27/2006. I will respond to your email in the order in which it was received. If you require immediate assistance, please call our toll free number, 888-227-5945, or email our general mailbox, [EMAIL PROTECTED] Thank You, Michael L. Young IT Manager [EMAIL PROTECTED] Administrative Claim Service, Inc. 888-227-5945 ___ --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 RTP Bridging
I know from everything in the past I have read, that Asterisk natively bridges calls between endpoints. We use * for only ACD and VMail purposes at this point, and I was wondering if there was any way to get a call from: PSTN-MGCP(cisco)-CCM-*(ACD)-Dial(SIP/)-CCM-(CCM phone) to not be bridged after the CCM connected phone answers. TIA for any help. -Greg ___ --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] FW: Exited non-zero
I am working on this app to dial two external numbers. The second is dialed
after the first hangs up. I have simplified things down to:
exten = 3852,1,Dial(zap/g1/3964,10,g)
exten = 3852,2,Wait(2)
exten = 3852,3,Dial(zap/g1/7757,10,g)
exten = 3852,4,Hangup
Here is the debug:
-- Accepting call from '00' to '3852' on channel 0/23, span 1
-- Executing Dial(Zap/23-1, zap/g1/3964|10|g) in new stack
-- Called g1/3964
-- Zap/1-1 is ringing
-- Zap/1-1 answered Zap/23-1
-- Attempting native bridge of Zap/23-1 and Zap/1-1
Everything is OK here when someone picks up x3964. Then when x3964 is hung
up, the first Dial command executes fine, returning -1 (as per the docs
since it is disconnected by the far end). This causes a debug message of:
-- Hungup 'Zap/1-1'
== Spawn extension (from-internal, 3852, 1) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
And execution halts.
I need to figure out either how to get the dial command to return 0 or get
the system to continue execution of the script despite the non-zero return.
Is there like an error handler / trap type routine that I can use in a
dialplan? I am going to try and dig through the source code of the dial
command, but there has got to be a better way...
Has anyone dealt with this before?
Thanks in advance,
Wes
-Original Message-
From: Schochet, Wes
Sent: Friday, January 13, 2006 2:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Bridging app
OK - this is great! However, I'm showing my lack of depth / newness here.
Calls from internal SIP phones work perfectly. Calls from external sources
(my PBX) fail. Obviously, I have a dialplan / context problem, but I'd
appreciate a brief explanation and some direction from the group!
In extensions.conf, I have [from-pstn]. Under that section, I have included
[ext-postcall]. Then I have the following in an included file:
[ext-postcall]
exten = 3852,1,Answer
exten = 3852,2,Dial(zap/g1/8030,10,g)
exten = 3852,3,wait(5)
exten = 3852,4,Dial(zap/g1/8041,10,g)
exten = 3852,5,wait(5)
exten = 3852,6,NoOp(${DIALSTATUS})
exten = 3852,7,Hangup
The trunk group g1 is a T1 to my PBX and in the zapata.conf file I have the
entry
context = from-pstn
..
..
Group = g1
Here is the trace from both an internal extension (205) and an external
extension.
From SIP/205 ( Context=from-internal in sip.conf ) asterisk*CLI
-- Executing Answer(SIP/205-1d7b, ) in new stack
-- Executing Dial(SIP/205-1d7b, zap/g1/8030|10|g) in new stack
-- Called g1/8030
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/205-1d7b
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing Dial(SIP/205-1d7b, zap/g1/8041|10|g) in new stack
-- Called g1/8041
-- Zap/1-1 is ringing
-- Zap/1-1 answered SIP/205-1d7b
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing NoOp(SIP/205-1d7b, ANSWER) in new stack
-- Executing Hangup(SIP/205-1d7b, ) in new stack
== Spawn extension (from-internal, 3852, 7) exited non-zero on
'SIP/205-1d7b'
-- Executing Macro(SIP/205-1d7b, hangupcall) in new stack
-- Executing ResetCDR(SIP/205-1d7b, w) in new stack
-- Executing NoCDR(SIP/205-1d7b, ) in new stack
-- Executing Wait(SIP/205-1d7b, 5) in new stack
-- Executing Hangup(SIP/205-1d7b, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/205-1d7b' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/205-1d7b'
asterisk*CLI
From External coming in Zap/g1 (from-pstn) :
asterisk*CLI
-- Executing Answer(Zap/23-1, ) in new stack
-- Accepting call from '00' to '3852' on channel 0/23, span 1
-- Executing Dial(Zap/23-1, zap/g1/8030|10|g) in new stack
-- Called g1/8030
-- Zap/1-1 is ringing
-- Zap/1-1 answered Zap/23-1
-- Attempting native bridge of Zap/23-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (from-pstn, 3852, 2) exited non-zero on 'Zap/23-1'
-- Executing Macro(Zap/23-1, hangupcall) in new stack
-- Executing ResetCDR(Zap/23-1, w) in new stack
-- Executing NoCDR(Zap/23-1, ) in new stack
-- Executing Wait(Zap/23-1, 5) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'Zap/23-1'
in macro 'hangupcall'
== Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/23-1'
-- Hungup 'Zap/23-1'
asterisk*CLI
-Original Message-
From: trixter aka Bret McDanel [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 12, 2006 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bridging app
There might be a simplier way. a channel variable that holds the users
response, and a gotoif. You should be able to pass 'g' to dial which
according to
[Asterisk-Users] MoH trouble with latest bristuff (0.3.0-PRE-1f)
Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan-timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but there was another issue, so I have to upgrade). Thanks in advance, Karsten ___ --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 Error 401 Problem
Dear All, I am having this strange problem on my Asterisk 1.2.1. We have a web dialer that can register to the Asterisk box in Hong Kong, but another user using the same account can't register to the Asterisk box using the same web dialer. Below is an output of the sip debug logs. It seems that the digest is missing the username and password, but why? I have also have this call flow for the an IP Phone, but after a while, it will register to the Asterisk. One thing I don't understand is that I have registered successfully in Hong Kong and when the user tries in South Africa, it doesn't work. Please Help! SIP Logs: From: sip:[EMAIL PROTECTED]To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERContact: *User-Agent: VaxSIP UserAgent/1.0Expires: 0Max-Forwards: 70Content-Length: 0 --- (11 headers 0 lines)---Using latest REGISTER request as basis requestSending to 192.168.0.3 : 2232 (non-NAT)Transmitting (NAT) to 196.38.228.123:5060:SIP/2.0 100 TryingVia: SIP/2.0/UDP 192.168.0.3:2232;received=196.38.228.123From: sip:[EMAIL PROTECTED]To: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYMax-Forwards: 70Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ---Transmitting (NAT) to 196.38.228.123:5060:SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 192.168.0.3:2232;received= 196.38.228.123From: sip:[EMAIL PROTECTED]To: sip:[EMAIL PROTECTED];tag=as63889026 Call-ID: [EMAIL PROTECTED]CSeq: 2 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70Contact: sip:[EMAIL PROTECTED]WWW-Authenticate: Digest realm=asterisk, nonce=4929aec7Content-Length: 0 Regards, Kengie ___ --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] making wakeup feature call phone number, not extension?
How would one go about setting up the wakeup feature of Asterisk to NOT call an extension, but to call a phone number? My setup works great for wakeup on local extensions, but I'd like to set it up to call external phone numbers automatically and play a specific sound file (to remind people of upcoming hair stylist appointments). I suppose either there'd have to be a web interface to use for this (entering a time for the reminder - and a phone number to call) or change the voice prompt to ask for a phone number to use, if not the extension called from. I'm sure it's doable - but I am now knowledgeable enough. I searched and didn't find any instructions on the web for something like this. 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] Echo
ping pong On 1/15/06, Steve Totaro [EMAIL PROTECTED] wrote: Just checking ___ --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] TE210P Trade
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I am stuck in a situation where I have two te110p's and really need a te210P. Anyone interested in a trade? They are in working condition guaranteed. If this is the wrong forum for this, I am sorry... I will take it off list. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDy/bny9wPyZpnL2URAp8NAJ9RVMrGwwFl/2khi+r/McQgSm8FZwCeNVzi qTHwUgTEFeizqMFJpK0oht8= =idMR -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
[Asterisk-Users] Support for RFC3323?
Does anybody know, if asterisk support the rfc 3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)?? I'm working with a Softswitch wich works with this rfc, and I don't know jet how to dissable this functionality. This is a problem becouse the SS do not pass the ANI in the interworking SIP-SS7 (only in this direcction). Regards Miguel ___ --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] loading zaptel drivers automatically upon reboot
On Sat, Jan 14, 2006 at 03:55:50PM -0500, Carlos Alperin wrote: After install everything on the supposedly right place, my conclusion is that zaptel doesn't load wct1xxp module. That's easy to test: before you restart zaptel, look at /proc/zaptel . if /proc/zaptel exists, zaptel was loaded . if /proc/zaptel/1 exists and reports those 24 channels, then wct1xxp has loaded and identified your card. Another possible reason: make sure that the zaptel init.d script runs before the asterisk one. It needs to have a lower start number. Use 'chkconfig --list asterisk' and 'chkconfig --list zaptel' to verify that. Then, that is the reason for Asterisk to fail loading. However I change the MODULES RMODULES on the zaptel on /etc/init.d /etc/sysconfig, it continuous same way. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Saturday, January 14, 2006 2:36 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] loading zaptel drivers automatically upon reboot On Fri, Jan 13, 2006 at 09:39:09PM -0500, Carlos Alperin wrote: That is right for zaptel. But you still has to do modprobe wctdm on rc.local before to load asterisk. rc.local is run after the standard init.d scripts. Thus if you load asterisk in an init.d script, you'd be loading the zaptel modules too late. Just add another init.d script. See the skeleton in /etc/init.d (there's a README there IIRC). -- 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 ___ --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 -- 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] I need feed back on how an Aastra VentureIP 4FXO
Well we have 3 sales people that are out visiting customers 50% or more of the time and it will get more as we got them laptops now. And if we can forward the calls to their cell phones with our phone system instead of giving the customers their cell numbers and then hanging up on the customer it will provide a better experience for the customer and better control for us. It may still be overkill but 4 lines aren't enough in the busy season and if we have 3 calling in and getting forwarded thru another to cells that is six already. And business is growing so we want room to expand before having to upgrade again. [EMAIL PROTECTED] wrote: 8 lines for 10 phones is overkillreally PaulH - Original Message - From: Tim Litwiller [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 15, 2006 2:38 PM Subject: Re: [Asterisk-Users] I need feed back on how an Aastra VentureIP 4FXO Thanks for the heads up - I didn't see anything that said it did work with asterisk so I thought I better ask. So if you where setting up a 6 - 8 telephone line system with 10 - 12 phones and trying to stay under $3000 for the system and phones what would you suggest. It sounds like if I can't do it for $3000 or under we will just stay with our old - outdated - partially functional phone system. I can probably reuse a workstation machine. And use AAH to make install and configuration easy. But that leaves some device ( suggest one to me) for * 8 fxo ports -And **12 voip phones * I think I'll just pass the fax/dsl line directly to the fax machine and dsl modem since we don't use it for anything else anyways and that means we don't have to worry about receiving faxes thru asterisk. ** I'd like to use the new Sipura 941's but may have to go with grandstream 2000's because of cost. We were supposed to have this done in November but cost issues have pushed it back this far already - so I'm not sure when this will happen. Cory Andrews wrote: The Aastra VentureIP system used a semi proprietary, non SIP protocol. I do not think it would integrate with Asterisk very well. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Carlos Chavez [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 14, 2006 6:04 PM Subject: Re: [Asterisk-Users] I need feed back on how an Aastra VentureIP 4FXO On Sat, 14 Jan 2006 11:22:51 -0600, Tim Litwiller wrote works with Asterisk. I'm thinking I'd need 2 to support 6-8 lines - Or suggest some other equipment that will provide up to 8 fxo ports and connect to asterisk. for future projects I'd also like something with 2 fxo ports and 4 - 5 fxs ports - I suppose a digium card would do fine for 2 fxo and 2fxs and I could do a sipura 2002 for 2 more. I do not think that the Venture IP will work with Asterisk at all. As far as I know it is a self contained system. The gateway unit will autoconfigure the phones so they work together. The firmware for the phones is not the same as the one used for SIP and Asterisk. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Monitor
Does anyone know of a web based live call monitor for *? I would have thought this was an ideal application for Ajax? TIA Simon ___ --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] uplink call quality issues
On Monday 16 January 2006 15:20, Esteban Guana-Jarrin wrote: We are using [EMAIL PROTECTED] 1.5 and SIP trunks to communicate to the PSTN network. We are having some problems with the call quality. Although we can hear the other person's voice quite clear when making or receiving a call, we get complaints from the people on the other end saying that our voices sound very unclear, low and that the voice drops, therefore people on the other end can not understand what we are saying. But as I said in our end their voices sound clear. I have checked network wise and found no latency problems within our small LAN, with our VoIP provider and reaching their SIP server's IP address, also the CPU load in the asterisk server has been graphed and does not exceed the normal CPU load levels Any assistance will be very much appreciated You could be saturating your upload traffic? What is the upload speed of you connection? hads -- Nap: Going back to sleep after taking a shower. -Jason ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problems with a pri (E1)
On 1/16/06, Xavier Gil [EMAIL PROTECTED] wrote: Hi all, Our asterisk PBX, randomly restarts all the channels of the E1 connection. It sends this message There is no D-Channel, using channel 16 anyway.Then the asteisk recive (or it thinks it recives) yellow alarms at all the B-channels, after that it restart all the channels. When restarting the B-channels it cut all the conversations that is handling at that moment. Does anyone have an idea for what it is happening? We run a asterisk 1.2.1 on a HP Proliant ML310, PIV 3Ghz 2 Mb L2 cache, 1 Gb Ram. We have a TE210P digium card configured for E1. This pbx has been running for almost a moth before giving this problems, we have called our telco and seens that in their side all is ok. (Our telco is ONO- Spain) This is the zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow bchan = 1-15,17-31 dchan = 16 # Global data loadzone= es defaultzone = es this is the zapata.conf [trunkgroups] [channels] language=es context=from-pstn switchtype=euroisdn pridialplan=unknown signalling=pri_cpe busydetect=yes callprogress=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no channel=1-15,17-31 callerid=asreceived thanks in advance, Xavier Gil Estarellas. Try adding resetinterval=never That may help. -- 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
Re: [Asterisk-Users] SIP RTP
On 1/14/06, Mike Hammett [EMAIL PROTECTED] wrote: According to this page: http://www.asterisk.org/doxygen/Config_sip.html canreinvite=yes redirects just the RTP. I was under the impression that the entire SIP connection got redirected, therefore losing accounting ability. Could someone clarify this? This isn't correct. While RTP goes away on a successful reinvite, Asterisk never gets out of the middle of the SIP signaling path because chan_sip is a B2BUA and not a SIP proxy. -- 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
Re: [Asterisk-Users] zaptel echo canceller preload patch
James Harper wrote: I've just posted the patch across to the dev list. I'll post it here if anyone asks for it, but didn't want to initially, lest the crossposting gods strike me down from above :) James As someone who doesn't appreciate what parts this effects, does this enhancement mean that after we : zt_ec_preload -d 1 echo_data_chan_1 that there would be no reason to use echotraining because this sets the channel up ahead of time and only changes caused by the echocancel's live adjusting would make any changes to the echocanceling settings/affect? Sorry if these are obvious questions, but I would like to know what I need to setup to properly test it ;). thanks, JES smime.p7s Description: S/MIME Cryptographic 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
[Asterisk-Users] Zapata.conf and Realtime
Sorry if this is a double-posting. I tried sending the following message Friday afternoon, but it still hasn't made it to the list. Based on the comments in the extconfig.conf file, zapata.conf *should* support being loaded realtime. Has anyone succeeded in doing so, and what does the schema, etc look like? Thanks! Steve ___ --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] ztdummy inaccuracy on linux-2.6
On Mon, 2006-01-16 at 17:43 +0100, Tamas wrote: Hello, I have some ugly numbers given by zttest for ztdummy on an AMD64 box running linux-2.6.15 compiled for Athlon64. linux-2.6.15, zaptel/branches/1.2 r900, jiffies ./zttest Opened pseudo zap interface, measuring accuracy... [snip] --- Results after 136 passes --- Best: 99.987793 -- Worst: 99.975586 -- Average: 99.975853 linux-2.6.15, zaptel/branches/1.2 r900, RTC Opened pseudo zap interface, measuring accuracy... [snip] --- Results after 96 passes --- Best: 99.963379 -- Worst: 99.938965 -- Average: 99.952942 linux-2.6.15, zaptel/branches/1.2 r900+patch bugs.digium.com/view.php?id=5971, RTC Opened pseudo zap interface, measuring accuracy... [snip] --- Results after 136 passes --- Best: 100.00 -- Worst: 99.694824 -- Average: 99.951973 HW: Tyan Tomcat K8E, Athlon64 3000+, 1GB RAM, 3ware 8006, 2x Maxtor HDD SW: Ubuntu 5.10, linux-2.6.15, zaptel from 1.2 branch Any idea what can be wrong? What does your /proc/interrupts say? On my asterisk box, I was seeing crappy interrupt handling like this only when I was using XT-PIC interrupt handling, when I moved to IO-APIC, things got much better... Steve ___ --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] realtime voicemail
On Sun, 2006-01-15 at 23:02 -0800, [EMAIL PROTECTED] wrote: i tried to setup realtime voicemail recently with 1.2.1 but couldn't get it to work. no matter what i do. it still looks for config in the voicemail.conf file. (BTW realtime sip extensions works fine) here's the voicemail line in extconfig.conf: voicemail = mysql,asterisk,voicemail here's the mysql schema: CREATE TABLE voicemail ( uniqueid int(11) NOT NULL auto_increment, customer_id bigint NOT NULL default '0', context varchar(50) NOT NULL default '', mailbox bigint NOT NULL default '0', password varchar(10) NOT NULL default '0', fullname varchar(50) NOT NULL default '', email varchar(50) NOT NULL default '', pager varchar(50) NOT NULL default '', stamp timestamp NOT NULL default CURRENT_TIMESTAMP on update CURRENT_TIMESTAMP, attach varchar(3) NOT NULL default 'yes', saycid varchar(3) NOT NULL default 'yes', hidefromdir varchar(3) NOT NULL default 'no', PRIMARY KEY (uniqueid), KEY mailbox_context (mailbox,context) ) TYPE=MyISAM; am i missing something? That looks like the minimal config... Something I found out was that I need to set the context = '' for it to work. I was using default, but for some reason I could never get that to work. Perhaps it is a bug? Steve ___ --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] List
On Mon, 2006-01-16 at 15:28 +1100, [EMAIL PROTECTED] wrote: The list is very quiet today - almost too quiet Yes, I have noticed the same thing. I have sent about 4 or 5 messages to the list, and the first one I sent (about 5 hrs ago) has yet to arrive. Perhaps there is something going on with the list-serv? Steve ___ --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] Random Disconnects
For me allowing reinvites is not an option, as Polycom phones with Sipura 3000 don't work together on this. On 1/16/06, Thczv F. Thczv [EMAIL PROTECTED] wrote: On 1/14/06, C F [EMAIL PROTECTED] wrote: I'm using Sipura 3000 as well, however I will have to wait until Monday about the Switch I'm not sure. So far it looks like Sipura is at fault. In the mean time I would like to hear from others using the Sipura 3000 FXO if they have the same problem. For now I am experimenting with allowing reinvites between the SPA-3000 FXO port and a couple of other extensions. I sent several inquiries to people who complained of this problem in the past (6+ months ago). I haven't heard back from them. 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] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
Since writing my message, I appear to have had success using iaxmodem + hylafax to do inbound faxing. Setup was not completely obvious, especially if you're usin [EMAIL PROTECTED], like me, I finally seem to have inbound faxes working properly now - 5 or 6 in a row have all come in just fine. Ben On 1/16/06, Philip Edelbrock [EMAIL PROTECTED] wrote: Ben Fried wrote: On 1/9/06, Rich Adamson [EMAIL PROTECTED] wrote: Sorry in advance if this is a FAQ... I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have a TDM400 card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM card. I haven't been able to get inbound fax with spandsp and rxfax to work. Occasionally an all-text fax will come in, though it's usually badly corrupted, but in most cases, it would appear that the call is terminated without successful transmission of the fax. I get logs that look what's included below. From reading the list, it looks like this is caused by the TDM card missing frames. Does that sound correct? If so, is there any relief in sight? Its been a problem since the card came out a couple of years ago. So, no it does not appear there is any relief in sight. Sigh. What a disappointment! Are there any other options for home users to receive faxes over the PSTN through *? Is anyone working on an alternative to the zaptel driver that might fix this issue? Humm, I tried to get my TDM400 card accepting faxes last week. It works about 1 out of 8 times. When it works, it looks great. When it doesn't, I usually (but not always) get a 'poor line quality' error from the sending fax machine and a blank or small corrupt image. I've tried adjusting the gain up and down, reversing ring/tip, and a few other little things. I wonder if it helps to adjust some other settings in the zapata.conf, like echo cancellation? My hope is dwindling, though, after reading this thread. :'( I do have a couple unused X100P clones sitting in that box that might be worth a try... 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
Re: [Asterisk-Users] ztdummy inaccuracy on linux-2.6
Steven Ringwald wrote: On Mon, 2006-01-16 at 17:43 +0100, Tamas wrote: Hello, I have some ugly numbers given by zttest for ztdummy on an AMD64 box running linux-2.6.15 compiled for Athlon64. linux-2.6.15, zaptel/branches/1.2 r900, jiffies ./zttest Opened pseudo zap interface, measuring accuracy... [snip] --- Results after 136 passes --- Best: 99.987793 -- Worst: 99.975586 -- Average: 99.975853 linux-2.6.15, zaptel/branches/1.2 r900, RTC Opened pseudo zap interface, measuring accuracy... [snip] --- Results after 96 passes --- Best: 99.963379 -- Worst: 99.938965 -- Average: 99.952942 linux-2.6.15, zaptel/branches/1.2 r900+patch bugs.digium.com/view.php?id=5971, RTC Opened pseudo zap interface, measuring accuracy... [snip] --- Results after 136 passes --- Best: 100.00 -- Worst: 99.694824 -- Average: 99.951973 HW: Tyan Tomcat K8E, Athlon64 3000+, 1GB RAM, 3ware 8006, 2x Maxtor HDD SW: Ubuntu 5.10, linux-2.6.15, zaptel from 1.2 branch Any idea what can be wrong? What does your /proc/interrupts say? On my asterisk box, I was seeing crappy interrupt handling like this only when I was using XT-PIC interrupt handling, when I moved to IO-APIC, things got much better... Steve cat /proc/interrupts CPU0 0:6645053IO-APIC-edge timer 1: 8IO-APIC-edge i8042 2: 0 XT-PIC cascade 5: 3309 IO-APIC-level eth1 7: 679362 IO-APIC-level eth0 8:8338011IO-APIC-edge rtc 10:204 IO-APIC-level eth2, HFC PCI 11: 20559 IO-APIC-level 3w- NMI:404 LOC:6644437 ERR: 0 MIS: 0 eth2 is not use currently. This box is in preparation for production. I don't know how can the HFC PCI card (Billion 1xBRI) get the same IRQ as eth2 [onboard Broadcom NIC]. Probably because it's on different bus: :04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express Flags: bus master, fast devsel, latency 0, IRQ 10 Memory at fe5f (64-bit, non-prefetchable) [size=64K] Capabilities: [48] Power Management version 2 Capabilities: [50] Vital Product Data Capabilities: [58] Message Signalled Interrupts: 64bit+ Queue=0/3 Enable- Capabilities: [d0] #10 [0001] :01:08.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Flags: bus master, medium devsel, latency 16, IRQ 10 I/O ports at d000 [disabled] [size=8] Memory at fdffc000 (32-bit, non-prefetchable) [size=256] Capabilities: [40] Power Management version 1 Anything else to take a look for? Thanks! Regards, Tamas ___ --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] Save the Quintum before I throw it out a window....
Well the subject line probably says it all. I have a Quintum D3000 which I'm supposed to be getting connected up to our Asterisk system. No matter what I try, neither username or authuser config works. I've also tried md5auth and it still refuses to register. Any one have a config they could share with me? Any help would be much appreciated. Neil ___ --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] Advice Of Charge (AOC) ?
On Sun, 15 Jan 2006, Patrick wrote: On Sun, 2006-01-15 at 10:32 +0100, Armin Schindler wrote: On Sun, 15 Jan 2006, Pisac wrote: Do Asterisk support Advice Of Charge (AOC) on ISDN lines? Do any ISDN drivers (bristuff, capi, vISDN, mISDN) support AOC? It is not implemented in chan_capi yet, but this is very easy. The question is what should be done with the AOC information? Just set some variable? As far as I know Asterisk has no API/structure for that. Hi Armin, Given the rather lackluster reaction to AOC so far I guess sticking it in a variable is the way to go. I don't know if KPN supports it. If they do and you decide to implement it then I'll be happy to test it for you. I can add this to chan_capi. Can you make a suggestion of the variable and the content you would expect? Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ztdummy inaccuracy on linux-2.6
On Mon, 2006-01-16 at 21:51 +0100, Tamas wrote: cat /proc/interrupts CPU0 0:6645053IO-APIC-edge timer 1: 8IO-APIC-edge i8042 2: 0 XT-PIC cascade 5: 3309 IO-APIC-level eth1 7: 679362 IO-APIC-level eth0 8:8338011IO-APIC-edge rtc 10:204 IO-APIC-level eth2, HFC PCI 11: 20559 IO-APIC-level 3w- NMI:404 LOC:6644437 ERR: 0 MIS: 0 eth2 is not use currently. This box is in preparation for production. I don't know how can the HFC PCI card (Billion 1xBRI) get the same IRQ as eth2 [onboard Broadcom NIC]. Probably because it's on different bus: :04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express (rev 11) Subsystem: Broadcom Corporation NetXtreme BCM5721 Gigabit Ethernet PCI Express Flags: bus master, fast devsel, latency 0, IRQ 10 Memory at fe5f (64-bit, non-prefetchable) [size=64K] Capabilities: [48] Power Management version 2 Capabilities: [50] Vital Product Data Capabilities: [58] Message Signalled Interrupts: 64bit+ Queue=0/3 Enable- Capabilities: [d0] #10 [0001] :01:08.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) Subsystem: Cologne Chip Designs GmbH ISDN Board Flags: bus master, medium devsel, latency 16, IRQ 10 I/O ports at d000 [disabled] [size=8] Memory at fdffc000 (32-bit, non-prefetchable) [size=256] Capabilities: [40] Power Management version 1 Anything else to take a look for? Ok. That looks like it *should* be working correctly. It is interesting that your cascade interrupt is still XT-PIC, and the highest interrupt listed is 11. I have attached the output from my /proc/interrupts for comparison. Does the bios have any mention of APIC/legacy or anything??? The board I am using is an Asus K8S-mx with a Sempron64 2800+ in it... Steve CPU0 0: 488733IO-APIC-edge timer 1:266IO-APIC-edge i8042 4: 4175IO-APIC-edge serial 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 169:1853920 IO-APIC-level libata, wct2xxp 177:492 IO-APIC-level eth0 185: 0 IO-APIC-level SiS SI7012 193: 9869 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217: 0 IO-APIC-level ohci_hcd:usb4 225:1850310 IO-APIC-level wcte11xp NMI:144 LOC: 488707 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe greeting message.
Ken D'Ambrosio wrote: So, I had it recorded _(as foo.gsm)_, modified app_meetme.c to reflect the Jan 16 12:53:05 WARNING[14859] file.c: Unable to open foo _(format ulaw)_: You recorded it in gsm, but it's looking for ulaw. Doug ___ --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 1.2.1 crashed
Hi guys, I'm using asterisk 1.2.1 since a week ago or so. today I found it crashed when making a call through teliax. This is how it looks: -- Called [EMAIL PROTECTED]/17075471770 -- Call accepted by 208.139.204.245 (format ulaw) -- Format for call is ulaw Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame -- IAX2/teliax-3 is making progress passing it to SIP/1010-9617 Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Jan 16 17:53:56 WARNING[5901]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0). $ the problem seems to be consistent. We crashed * several times. A mail was sent to me by safe_asterisk telling it exited with signal 11. And I have 4 core files (each one for each crash we experimented today) if they are useful. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
