[Asterisk-Users] Making asterisk distributed
Hi, I want to know, if someone has tried to use clustering in asterisk to increase its scalability and make it distributed?? If yes, how easy it is to cluster? Can someone please ive me details about the same Thanks Varun __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec+packet loss concealment
Hi From what I have heard, Asterisk does not allow for iLBC to take advantage of the lost packet concealment. I understand this has something to do with the jitter processing. If lost packet concealment doesnt work with ilbc, I can assume the same applies to other codecs who claim to have this feature. Hopefully this will be fixed sometime soon, especially for us folks with less than ideal IP throughput. Regards Clive On Tue, 03 Aug 2004 10:22:20 +1000 Adam Hart [EMAIL PROTECTED] wrote: Steve Underwood wrote: Adam Hart wrote: Daniel Niasoff wrote: Is G729 more sensitive to packet loss or delays due to its higher compression. If Ive generally got the bandwidth available, am I best sticking to ulaw. G.729 has lost packet concealment, G.711 doesn't. G.711 will sound better otherwise if you can afford the bandwidth. Eh? G.729 has no particular features to allow more effective packet loss concealment. iLBC has, but at the cost of a substantially higher bit rate. In fact G.711 is a little ahead of G.729 in the regard, since packets are completely independant. The smoothing in G.729 means you need the previous packet to decode the current one properly. Regards, Steve I believe you're mistaken - G.729 works similar to iLBC and speex. iLBC works better as the packets are independent but G.729 still has a function for packet loss concealment. prehaps have a look at http://www.speex.org/comparison.html -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Herbalife Independent Distributor http://www.healthiest.co.za ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Making asterisk distributed
Trilogy India wrote: Hi, I want to know, if someone has tried to use clustering in asterisk to increase its scalability and make it distributed?? If yes, how easy it is to cluster? Can someone please ive me details about the same Thanks Varun This has been discussed a number of times in the past, searching the archives will reveal the various technical reasons why a clustered SIP solution is very difficult to implement.. Your only real option for scalability is to distribute your users over many servers and interlink them with IAX and your dial plan.. For those who don't know how to search the archives.. Goto the bottom of the page at http://www.digium.com/index.php?menu=mailing_list and search.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec
On Tue, 3 Aug 2004, Steve Underwood wrote: Eh? G.729 has no particular features to allow more effective packet loss concealment. iLBC has, but at the cost of a substantially higher bit rate. In fact G.711 is a little ahead of G.729 in the regard, since packets are completely independant. The smoothing in G.729 means you need the previous packet to decode the current one properly. For IAX2, at least, Asterisk oes not use the lost-packet-concealment of any codec. This is because the incoming frames clock Asterisk. For iLBC's lost packet concealment to work, Asterisk would have to start calling the decoder with a NULL at the point when the missing packet shold have arrived. Can't say for sure for SIP, but I'd guess that its the same. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing
Thanks for help. All works now. Problem was in codecs on different sides Definity: display ds1 1b14 CRC? n Interface Companding: mulaw And when making call via asterisk Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law ^ (35) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 So I can't make call. But incoming call (Definity - Asterisk) works, because asterisk understand ulaw. So, I have once more question. How can I change codec on Digium card on Asterisk side? I configure asterisk and definity with this page (http://www.voip-info.org/wiki-Asterisk+Avaya), and here no one word about what codec asterisk use. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 not dial Modem[i4l]/g1
Hello everybody,
I have a strange comportment with oh323 and asterisk, I'start testing
asterisk but with this I can't understant plesae help me !
Thanks
Eltorio
--
1/PB: I can't dial from a H323 extensions (registered on a GNU GK) to a
Modem[i4l] line
--
Nothing happens when h323 to isdn even everything is ok for isdn to SIP and
isdn to h323 and sip to isdn!: trace
--
2/ versions used
--
Asterisk CVS-HEAD-07/29/04-19:00:52 built by [EMAIL PROTECTED] on a i686 running
Linux
Linux compiere 2.6.4-52-smp #1 SMP Wed Apr 7 02:11:20 UTC 2004 i686 i686
i386 GNU/Linux (Suse 9.1)
Oh323 0.6.3
---
2-1 test FAILED NetMeeting(4001 on GnuGK) To ISDN Phone 221 (strip in phone
1 digit) so 5221 is dialed 221
--
-- Executing Dial(OH323/R21368, Modem/g1:5221) in new stack
-- Called g1:5221
(after more than 1 minute I stop Netmeeting call)
-- H.323 call 'ip$192.168.3.1:30056/21368' cleared, reason 7 (Remote
user stopped calling)
-- Hungup 'Modem[i4l]/ttyI1'
== Spawn extension (lesmuids, 5221, 1) exited non-zero on 'OH323/R21368'
-- Hungup 'OH323/R21368'
--
2-2 test OK SIP Phone (on Asterisk) call 5221
--
-- Executing Dial(SIP/4122-d20d, Modem/g1:5221) in new stack
Aug 3 02:54:22 WARNING[1141783472]: chan_modem_i4l.c:608 i4l_dial:
Outgoing MSN 4122 not allowed (see outgoingmsn=,, in modem.conf)
-- Called g1:5221
-- Modem[i4l]/ttyI1 answered SIP/4122-d20d
(blah blah.. and hang)
-- Hungup 'Modem[i4l]/ttyI1'
== Spawn extension (lesmuids, 5221, 1) exited non-zero on 'SIP/4122-d20d'
Strange think I can call a SIP extension
--
2-3 test OK ISDN phone NetMeeting via Asterisk
--
-- Executing Goto(Modem[i4l]/ttyI0, 4001|1) in new stack
-- Goto (lesmuids,4001,1)
-- Executing Dial(Modem[i4l]/ttyI0, OH323/4001) in new stack
-- H.323 call to 4001 with codec ALAW
-- Called 4001
-- OH323/L20707 is ringing
-- OH323/L20707 answered Modem[i4l]/ttyI0
-- Hungup 'OH323/L20707'
== Spawn extension (lesmuids, 4001, 1) exited non-zero on
'Modem[i4l]/ttyI0'
-- H.323 call 'ip$localhost/20707' cleared, reason 1 (Cleared by local
user)
(blah blah)
-- Hungup 'Modem[i4l]/ttyI0'
--
3 Config
--
Extensions.conf
[lesmuids]
;Accueil application (sur SIP/0, ou sur ISDN/400)
exten = 0,1,Wait,15; Attend 15s (4 eme
sonnerie)
exten = 0,2,NoOp(Receive on 0)
exten = 0,3,Dial(SIP/4122SIP/4123,60,tr)
exten = 1,1,Goto(4122,1)
exten = 2,1,Goto(4122,1)
exten = _[3-9],1,Goto(4001,1)
exten = 4001,1,Dial(OH323/4001)
exten = 4002,1,Dial(OH323/4002)
exten = 4008,1,Dial(OH323/408)
exten = 4122,1,Dial(SIP/4122,60,tr)
exten = 4123,1,Dial(SIP/4123,60,tr)
exten = _[5]ZXX,1,Dial(Modem/g1:${EXTEN})
oh323.conf
[general]
listenAddress=0.0.0.0
listenPort=1720
connectPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=no
h245Tunnelling=no
h245inSetup=no
inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
bandwidthLimit=1024
wrapLibTraceLevel=1
libTraceLevel=0
libTraceFile=stdout
gatekeeper=DISCOVER
gatekeeperTTL=100
userInputMode=TONE
amaFlags=omit
context=lesmuids
[register]
context=lesmuids
gwprefix=41
gwprefix=5
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[Asterisk-Users] Using Clustering/TDMoE
Hi, I want to know how we can use TDMoE to cluster asterisk?? And, how many asterisk servers it can cluster and how?? Varun __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Clustering/TDMoE
On Tue, 2004-08-03 at 03:43, Trilogy India wrote: Hi, I want to know how we can use TDMoE to cluster asterisk?? And, how many asterisk servers it can cluster and how?? If you want to know how, search the archives, or read the wiki. Once you have specific questions instead of blanket implementation, come back and ask the specific question. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] App.c
On Mon, 2 Aug 2004 12:32:38 -0400, AJ Grinnell [EMAIL PROTECTED] wrote: Can someone tell me where I can get just app.c from. Mine somehow got corrupted, and no updates or anything else will fix it. I just need the one file from the latest cvs. 8-1-04. Please help Delete your corrupted app.c and re download from cvs Then make clean make install Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as Front-End for Artisoft Televantage 6
On Mon, 2 Aug 2004 12:54:59 -0700, Alain Bautista [EMAIL PROTECTED] wrote: Anyone had experience 'marrying' the two? We had setup * to front end Artisoft's Televantage. It works with some issues need to be resolved: - Inbound calls could not properly handled and routed by Televantage's Call Classifier. It goes directly to the Televantage's default auto Some more information on how the two systems are connected would help are you using PRI, T1, Analogue etc... Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco PRI no CallerID
On Mon, 02 Aug 2004 20:23:24 -0700, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: * -- SIP -- CISCO -- PRI -- PSTN The PSTN sees no callerid. *--- PRI[zaptel]-- PSTN Callerid is there... which makes me think it's the cisco, not the PRI/PSTN/telco. CISCO PRI-- * PRI [zaptel] Callerid IS there... which makes me shake my head in disbelief, because * can see clid from the cisco pri, but pstn doesn't... but when * sends info on that pri, pstn does see clid. help? It sounds like your Carrier is blocking the CLI on it's PSTN there is nothig you can do about it but talk to them. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco PRI no CallerID
On Mon, 02 Aug 2004 20:23:24 -0700, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: * -- SIP -- CISCO -- PRI -- PSTN The PSTN sees no callerid. *--- PRI[zaptel]-- PSTN Callerid is there... which makes me think it's the cisco, not the PRI/PSTN/telco. CISCO PRI-- * PRI [zaptel] Callerid IS there... which makes me shake my head in disbelief, because * can see clid from the cisco pri, but pstn doesn't... but when * sends info on that pri, pstn does see clid. help? A lot of carriers do CLI validation but it may also be as simple as the numbering plan/type that you are sending on outbound ISDN calls. Your carrier should of specified how they would like to receive the CLI (national/international format/preceeding zero maybe). As Jason said check with your carrier ... * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] avm c4, ptmp
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 02 August 2004 18:39, Deti Fliegl wrote: Your Extension has to match your MSNs. You have to configure all MSNs you have in a comma separated list like msn=27849,27852,27869,27861 and you must only use these MSNs as caller id. Hi :) thnx for having tryied to help :) we have 2 number on our isdn: 0721855285 and 0721859609 i try to call my home: 0721950396 here the issue: now in capi.conf i've: # cat capi.conf ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] [controller1] msn=0721855285,0721859609 incomingmsn=* controller=1,2,3,4 softdtmf=1 accountcode= context=default ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 mode=immediate isdnmode=p2mp ; ;-- in extension.conf i have: [local] ignorepat = 9 exten = _9XX.,1,Dial,CAPI/0721855285:bBYEXTENSION:1 exten = _9XX.,2,Congestion exten = _9XX.,3,Hangup Aug 3 11:26:31 DEBUG[1145346992]: chan_sip.c:4423 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 -- Executing Dial(SIP/sip1-5fcd, CAPI/0721855285:bBYEXTENSION:1) in new stack -- data = 0721855285:b90721950396:1 -- capi request omsn = 0721855285 Aug 3 11:26:31 NOTICE[1224625072]: chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn = 0721855285. you should check your config! Aug 3 11:26:31 NOTICE[1224625072]: app_dial.c:714 dial_exec: Unable to create channel of type 'CAPI' as yuo can see, -- data = 0721855285:b90721950396:1 -- capi request omsn = 0721855285 everithing seems ok :) byez Maurizio - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBD12N4Q/49nIJTlwRApRtAJ94VfuG+F00IJRuyIz7vbajjLOmggCfcAwT RFhrkbzXB3TBqcieHz5k74A= =Pms4 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk+radius
HI ALL; Is there anybody who use app_radius(astersik radius module)??? is it stable? Regards mohammad
Re: [Asterisk-Users] G729 Codec+packet loss concealment
On Tuesday 03 August 2004 02:58, [EMAIL PROTECTED] wrote: From what I have heard, Asterisk does not allow for iLBC to take advantage of the lost packet concealment. I understand this has something to do with the jitter processing. Can you provide a source for that statement? I am not disputing it but I'd like to have it in the archives for one, but also to verify the claim too. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] avm c4, ptmp
On Tue, 3 Aug 2004 11:40:28 +0200, Maurizio Marini [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 02 August 2004 18:39, Deti Fliegl wrote: Your Extension has to match your MSNs. You have to configure all MSNs you have in a comma separated list like msn=27849,27852,27869,27861 and you must only use these MSNs as caller id. Hi :) thnx for having tryied to help :) we have 2 number on our isdn: 0721855285 and 0721859609 i try to call my home: 0721950396 here the issue: I would set the MSN's to 855285 and 859609 They do not usually include the area code. Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] avm c4, ptmp
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jason :) On Tuesday 03 August 2004 12:07, Jason Williams wrote: I would set the MSN's to 855285 and 859609 They do not usually include the area code. [local] exten = _9XX.,1,Dial,CAPI/855285:bBYEXTENSION:1 exten = _9XX.,2,Congestion exten = _9XX.,3,Hangup ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] [controller1] msn=855285,859609 incomingmsn=* controller=1,2,3,4 softdtmf=1 accountcode= context=local ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 mode=immediate isdnmode=p2mp ; ;-- Aug 3 12:02:28 DEBUG[1145346992]: chan_sip.c:4423 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060 -- Executing Dial(SIP/sip1-0167, CAPI/855285:bBYEXTENSION:1) in new stack -- data = 855285:b90721950396:1 -- capi request omsn = 855285 Aug 3 12:02:28 NOTICE[1224625072]: chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn = 855285. you should check your config! Aug 3 12:02:28 NOTICE[1224625072]: app_dial.c:714 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy/congested at this time - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBD2W24Q/49nIJTlwRAi0cAJ4/ckdwqJMDbWVYYsMU8wj9zksbugCeJfl5 lh2CHTrKNg7WOhqfFf/B1Zo= =LVNs -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo problems with mISDN?
Hi all, Im using a AVM Fritz! Card (TE-Mode) and a Longshine LCS-8051 with HFC-S-Chip (NT-Mode) together with the chan_misdn. I build the system like it was explained at http://isdn.jolly.de. At first I used the pbx4linux software from jolly (http://isdn.jolly.de) and then I changed to asterisk. Both systems works fine, but I have (in both systems) the same echo problem. When I call a person (ISDN to ISDN), I have an echo, and the other person has no echo! But when I call to a cell phone (ISDN to GSM), I dont have an echo! So this works fine. How can I get rid of this echo? Maybe I have to buy another card? Or is there something wrong with the mISDN drivers? Regards, Christian
[Asterisk-Users] Called ID in Australia
Hello All, Can any Australians who have any info or current patches relating to Caller ID in Australia please drop me a line? There is little or no info on the Wiki regarding this topic, although I am aware of a related patch mentioned in the bug tracker. Regards, Rob Barnes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp2 testers needed
On Fri, Jul 30, 2004 at 03:40:58AM +0200, Jan Czmok wrote: Date: Fri, 30 Jul 2004 03:40:58 +0200 From: Jan Czmok [EMAIL PROTECTED] Dear Skinny/SCCP lovers :-) I've just completed uploaded to the cvs the newest version with fixed redial key AND implementation of speed dials. please test extensively and report any bugs. i know that the display is not yet set correctly but the buttons are working as expected. Enjoy testing... Is it possible to make chan_sccp work with Cisco's SRST feature? Chan_sccp complains: chan_sccp.c:134 handle_message: Client sent RegisterTokenReq without first registering. when phone tries to register form call-manager-fallback (SRST) router back to Asterisk. Thanks in advance. -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Called ID in Australia
Rob, Caller ID all depends on which hardware you're using. I can say that if you're using chan_capi (for CAPI compatible ISDN hardware) caller ID works perfectly. You'll find getting it working is highly dependant on which hardware and therefore channel driver you're using. Regards, Kimble Young -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Barnes Sent: Tuesday, August 03, 2004 8:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Called ID in Australia Hello All, Can any Australians who have any info or current patches relating to Caller ID in Australia please drop me a line? There is little or no info on the Wiki regarding this topic, although I am aware of a related patch mentioned in the bug tracker. Regards, Rob Barnes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help with digium E1 card
A couple things: In zapata.conf, the channels line should be: channel = 1-15,17-31 Thanks, now asterisk loads without error. If you are connected to the PSTN, the signalling should be pri_cpe (customer premise equipment). But your setting would be correct for connection to a channel bank I think. I'm testing connected to Cisco AS5300. Hope this helps Scott Gracias! HoraPe --- Horacio J. Peña [EMAIL PROTECTED] [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A few questions
Hi All, As a company, we are looking to rationalize our phone system infrastructure and have come across using a digium quad port E1 PRI cards in conjunction with the Asterisk PBX software. I'm hoping you'll be able to answer the following questions and maybe give me a few configuration hints. Presently I have an Asterisk installation using a Fritz card and a BRI line for testing, and unfortunately, I don't have any DDI's configured on the BRI line. We have several C/T servers with PRI lines that are under utilised, in the following configuration eISDN - PRI - C/T Server 1 eISDN - PRI - C/T Server 2 eISDN - PRI - C/T Server 3 We wish to use Asterisk as a switch to direct calls (based on the dialled number) to the correct C/T server - PRI - C/T Server 1 / eISDN - PRI - Asterisk -- PRI - C/T Server 2 \ - PRI - C/T Server 3 For our C/T applications we need the Dialed Number passing from the PRI to the C/T server - is this possible ? If we install 2 or more of the Quad port ISDN cards, and a call came in on the first card, but was re-directed out of a second card, is there a dedicated bus between the cards (as with Dialogic cards) or would it use the Server's PCI bus ? Do you have any idea of the extra load this would put on the CPU ? We also have a Samsung DCS phone switch that connects to 4 BRI lines, do you have or know of any product that will work with asterisk and allow us to connect this to the Asterisk server ? Ie FROM:- eISDN 4xBRI - DCS To :- Asterisk - 4xBRI - DCS Thanks in advance for any information Mark Wilkinson 2PM Technologies Ltd. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????
On Mon, 2004-08-02 at 20:32, Bartosz Wegrzyn wrote: I joined this group 2 weeks ago, because I was having problems with my asterisk box and broadvoice. I found many discussions regarding similar issue. I belive that this is the group where we can share our problems and help each other. We know that broadvoice does not suport the asterisk, and the only person who can help is James Jones and you asterisk people. I cannot understand why you are so angry about my post. I don't know what kind of computer GURU you are, but I am a regular networking person who want to have the things up and running. Maybe, the way I presented the facts was not that professional as you do, but this is how I do it. Maybe you should just look at the post and try to help. It isn't so much how you have brought this question. It is just that less than 1% of the subscribers of this list actually have even the slightest of interest in broadvoice. When the traffic for such a small percentage of the list gets to be 25-40% of the traffic, most of us start getting edgy and wishing the traffic was somewhere else. No offense, Steve, but the exact same words apply to h323, odbc, isdn cards, Nufone, etc, for a lot of the rest of us. However, historically we've simply been deleting posts of no interest (to many, but certainly not all). After many many years of using various Internet resources, it is simply amazing how many people on this _user_ list try to re-define the purpose of the list (to their liking), exactly how people should post (to make _their_ reader more convenient for them), etc. What has been rather interesting is the broadvoice change seems to indicate there is a * sip dns issue that has not yet been sufficiently documented to enter a bug report. (Example: given a type=user and type=peer sip context, broadvoice incoming calls actually use the type=peer context. What?) So, for those that don't have any interest in the broadvoice interface topic, find your delete key. Its not all that hard, really. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec+packet loss concealment
On Tue, 3 Aug 2004, Andrew Kohlsmith wrote: On Tuesday 03 August 2004 02:58, [EMAIL PROTECTED] wrote: From what I have heard, Asterisk does not allow for iLBC to take advantage of the lost packet concealment. I understand this has something to do with the jitter processing. Can you provide a source for that statement? I am not disputing it but I'd like to have it in the archives for one, but also to verify the claim too. I am the source for that statement. Is that a problem? ;-) My qualification is having worked on the IAX2 jitter buffer, consequently having studied how audio flows from the received frames through the jitter buffer and then via ast_translate() into the codec. To use the iLBC codec's lost packet concealment, you must call the codec every 20msec (or whatever). If you have a new frame, you pass it - and you get that back decoded. If you haven't, you call iLBC_decode anyway, passing mode=0 and get a reconstructed frame back. In Asterisk, bridging is self-clocked by the incoming frames. So unless a frame arrives, nothing happens. Each arriving frame gets pushed through the codec decode function. But if a frame doesn't turn up... well, then... nothing will happen. You can confirm this by examining codecs/codec_ilbc.c where you will see that there is only one call to iLBC_decode(), and that call has a hardcoded mode=1. So Asterisk will never use the iLBC packet loss concealment capability. At the moment, anyway - this can arguably be fixed, but its not a trivial fix. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Called ID in Australia
The only change I believe I had to make was under /usr/src/asterisk/channels/chan_zap.c #define DEFAULT_CIDRINGS 2 The default is 1 Google search if you want some of the previous threads... http://www.google.com.au/search?q=asterisk+callerid+patch+australiaie=UTF-8 hl=enmeta= -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Robert Barnes Sent: Tuesday, 3 August 2004 8:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Called ID in Australia Hello All, Can any Australians who have any info or current patches relating to Caller ID in Australia please drop me a line? There is little or no info on the Wiki regarding this topic, although I am aware of a related patch mentioned in the bug tracker. Regards, Rob Barnes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec+packet loss concealment
On Tuesday 03 August 2004 07:18, [EMAIL PROTECTED] wrote: I am the source for that statement. Is that a problem? ;-) Not at all. :-) But I do thank you for taking the time to write a few paragraphs explaining what's going on in the current code. It's certainly something I didn't know before and it really takes out an advantage of using iLBC over something like GSM -- the former has lower bandwidth requirements but the increased processor overhead and lag might be causing another problem I've been seeing. I've switched back to GSM for now to see if the audio quality issues goes away. My qualification is having worked on the IAX2 jitter buffer, consequently having studied how audio flows from the received frames through the jitter buffer and then via ast_translate() into the codec. Hmm... having worked on the IAX2 jitter buffer, can you tell us why trunking and jitter buffers don't get along? When trunking with nufone I get ... interesting... audio if I have a jitter buffer enabled. :-) Hey there, how are you today turns into Heytherehowareyoutoday In Asterisk, bridging is self-clocked by the incoming frames. So unless a frame arrives, nothing happens. Each arriving frame gets pushed through the codec decode function. But if a frame doesn't turn up... well, then... nothing will happen. Aside from easier implementation is there any advantage to having the audio streams self-clocked? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any small colleges/universities using PBX or Voicemail?
What an ACTIVE newsgroup! I'm in the early stages of researching Asterisk. My current environment is a small college (~1000 sets/~400 student sets), Avaya Definity G3si/Seimens Rolm Phonemail. As you can imagine, the maintenance, licensing, and equipment costs are HEFTY. So.. are there any small colleges/universities using PBX or Voicemail? If so, I'd be interested in your migration path. What equipment was replaced, and how did you handle the loss of investment in any proprietary sets? Many thanks, Brian Hudson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Called ID in Australia
On Tue, 3 Aug 2004 20:24:50 +1000, Robert Barnes [EMAIL PROTECTED] wrote: Hello All, Can any Australians who have any info or current patches relating to Caller ID in Australia please drop me a line? There is little or no info on the Wiki regarding this topic, although I am aware of a related patch mentioned in the bug tracker. Regards, Rob Barnes Australia uses the bellcore caller ID standard which is the same as that used in the USA, Canada, and a few other countries. Depending on the hardware you use, caller ID should work out of the box - at least it did with my ISDN and X100P cards. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any small colleges/universities using PBX or Voicemail?
On Tue, 2004-08-03 at 08:21, Brian Hudson wrote: What an ACTIVE newsgroup! I'm in the early stages of researching Asterisk. My current environment is a small college (~1000 sets/~400 student sets), Avaya Definity G3si/Seimens Rolm Phonemail. As you can imagine, the maintenance, licensing, and equipment costs are HEFTY. So.. are there any small colleges/universities using PBX or Voicemail? If so, I'd be interested in your migration path. What equipment was replaced, and how did you handle the loss of investment in any proprietary sets? Many thanks, Brian Hudson Brian, check the last few days of the list - several people have been talking about integrating systems like yours and asterisk. Remember the wiki is your friend! http://www.voip-info.org/wiki-Asterisk+Avaya t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] avm c4: DISCONNECT_IND ID=001 #0x0193 LEN=0014
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 i fixed wrong capi.conf (there was [controller1] after [interfaces]) now capi.conf is: ; ; CAPI config ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=855285,859609 incomingmsn=* controller=1,2,3,4 softdtmf=0 accountcode= context=local ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=4 mode=immediate isdnmode=ptm question: devices should be 4 or 2? now when a issue a call i get: Aug 3 14:43:45 DEBUG[1224625072]: pbx.c:1255 pbx_extension_helper: Launching 'Dial' -- data = 855285:0721950396 -- capi request omsn = 855285 == found capi with omsn = 855285 Urgent handler == CAPI Call CAPI[contr1/855285]/6 -- Called 855285:0721950396 Urgent handler -- CONNECT_CONF ID=001 #0x0010 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 -- CONNECT_CONF ID=001 #0x0010 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 Urgent handler Aug 3 14:43:45 DEBUG[1224625072]: channel.c:1699 ast_set_read_format: Set channel CAPI[contr1/855285]/6 to read format ULAW Aug 3 14:43:45 DEBUG[1224625072]: channel.c:1666 ast_set_write_format: Set channel SIP/sip1-9316 to write format ULAW Aug 3 14:43:45 DEBUG[1224625072]: channel.c:1666 ast_set_write_format: Set channel CAPI[contr1/855285]/6 to write format ALAW Aug 3 14:43:45 DEBUG[1224625072]: channel.c:1699 ast_set_read_format: Set channel SIP/sip1-9316 to read format ALAW -- DISCONNECT_IND ID=001 #0x0193 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3301 == DISCONNECT_IND PLCI=0x101 REASON=0x3301 Urgent handler someone knows? 10x - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBD40e4Q/49nIJTlwRAnq/AJ0dJ3ybyYOlh8xtQYDdvS4xT3BNLwCeN74p r7OJfCwcpDqccyKq1S+YWXA= =pfSZ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fw: [Asterisk-Users] Digium FXO Interfaces don't support groundstart???
A thought occurred to me to on how to further quantify the impact of glare on a properly dimensioned trunk group and debunk the ground start glare concern. A cursory traffic analysis clarifies: 1. Assume you have a two-way trunk group, dimensioned for average busy hour, average busy season for P01 grade of service, Erlang B or Poison distribution. 2. Further assume that the average holding time per call is the national average of 300s (5 minutes). 3. Further assume that you have a worse case CPE incoming glare condition by using the previously described brain dead arrangement of an ancient PBX with ring detect as an incoming connect signal and a worse case CO that has non-immediate ring (2s on 4s off cadence), yielding a horrific average glare interval of 2 seconds instead of the typical of 10-50ms that would be normal for change of state detection/immediate ring typical loop supervision line. 4. Further assume that the two-way hunt order is inverse (e.g., CO ascending, CPE descending terminal hunt) as it should be. What would be the expected incidence of glare in the busy hour? By definition (1, above) the probability that any attempt arriving for the assumed offered load, arriving at random, having exponentially distributed holding times and having infinite sources will find any circuit busy would be 1 out of 100 busy hour attempts for all calls (including those that encounter glare). Understand first that only 1 out of 100 attemps will find any facility occupied at all. Then, for the busy event that is encountered, what is the probability that it is the result of two calls, arriving at random, and seize the same trunk from opposite ends during a concurrent 2 second window for the assumed traffic intensity? Intuitively, encountering a glare condition would certainly have to be far less probable than encountering just any busy condition, which is the traffic distribution equation's prediction. In fact, since the average call holding time of 300 seconds (5 minutes) is 150 times more than the assumed average glare interval of 2 seconds, it is far more likely to find a call in the non-glare state when the predicted loss event occurs than finding it is the result of the assumed glare condition of two calls, arriving at random from opposite ends, occuring within a window of the 2 seconds. This is not just a simple ratio of the average glare holding time versus the average call holding time (because the distribution equation assumes exponential distribution of holding times), but it is far less than 1 out of 1000 attempts for the assumed average holding time and the average glare interval. To put some real numbers on this for illustration, a trunk group for a full T1 of 24 channels will carry 21.125 erlang (Poisson) in the busy hour. It will have 253 attempts (from the typical average holding time assumed in 2, above), of which only three will encounter a busy. Of the three that encounter a busy, it is very unlikely any would encounter glare. Over time (an interval of days), some nevertheless will. Note that this is not what was previously described in the horror stories related. Trunks were sticking all over the place and a rash of problems were encountered, provoking panic. The situation was so awful that it caused the ensuing foaming at the mouth, displacing it to a fault of Digium being too cheap to put in required features, or ignorant engineers for not having the wisdom and experience to insist on ground start to alleviate the near certain disaster that would inevitably occur on both large or small trunk groups. Now let's take a look at a far more likely scenario that is supported by the same traffic theory: 5. Assume that you have both ends hunting in the same order from both directions, i. e., the same hunt order from both the central office and the PBX (say both hunt in ascending order). The remaining assumptions (1 through 3, above) still are valid. The traffic scenario drastically changes. Now the incoming/outgoing traffic is always focused on the next idle line instead of the full dimension of all of the servers in the group. In essence it behaves like a trunk group of 1 and the probability of failure now rises asymptotically. Two hundred fifty-three busy hour attempts (from the previous example) on a trunk group of one would raise the incidence of failure due to glare significantly and the now trivial 2 seconds (or even 50ms) is of dire importance. Worse, the glare interval is not over several days and confined to the busy hour, it now raises its ugly head for just about any time of the day. But it becomes massive during the busy hour. Trunks hang, subscribers complain, and the situtation looks like it is going to hell in a hand basket. Ground start is invoked and the problem appears to go away. The ground start cure simply masks the problem. If a identical hunting arrangement is used at both ends, PBX and CO 2nd trial failures abound, but calls do go
[Asterisk-Users] VON Magazine article.
Just opened my July/August VON Magazine, and as usual started reading from the back. SIP at Risk and Asterisk caught my eye, gives */IAX a nice plug. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Called ID in Australia
On Tue, 2004-08-03 at 21:30, Christopher Lee wrote: The only change I believe I had to make was under /usr/src/asterisk/channels/chan_zap.c #define DEFAULT_CIDRINGS 2 The default is 1 This is one of 2 patches I make to asterisk every time I download. It is needed to make the callerid appear on an australian handset connected to an FXS port. I suppose if you used SIP/etc then it wouldn't be needed. I haven't tested an FXO port, but I assume the same change would be needed. It would be nice if this could be set from zapata.conf rather than in the source code Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configure Makefile to run with older iax Protocol
Hi guys, ive heard that the latest version of asterisk can be compiled to run with the old iax1 protocol as a default. Any ideas ? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] features.conf
Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf
-Original Message- From: AJ Grinnell [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 10:28 AM To: Asterisk Subject: [Asterisk-Users] features.conf Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it? I think that it should be in configs/features.conf.sample unless you have run make samples in which case this file is copied to /etc/asterisk. Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astguiclient: blank php pages
Hi, I just installed astguiclient, following the SCRATCH_INSTALL, without errors. But when I try to enter the administration page (http://127.0.0.1/astguiclient/ admin.php), it's blank. The browser shows me the following page source: htmlbody/body/html The same happens with http://127.0.0.1/astguiclient/welcome.php What could be wrong? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] astguiclient: blank php pages
There could be several causes for this. First, check your php.ini file to see that Globals are turned on. Did you do a full install from scratch and follow the instructions from the beginning? We can continue this off-list as to not annoy everyone with troubleshooting. MATT--- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 10:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] astguiclient: blank php pages Hi, I just installed astguiclient, following the SCRATCH_INSTALL, without errors. But when I try to enter the administration page (http://127.0.0.1/astguiclient/ admin.php), it's blank. The browser shows me the following page source: htmlbody/body/html The same happens with http://127.0.0.1/astguiclient/welcome.php What could be wrong? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID Trunk
I'm working on putting together some Ideas about using Asterisk in our environment, one of the things I want to consider is DID trunks (analog), what hardware do I need to terminate these trunks? I'm looking at the voicetronix openswitch6 or openswitch12. On the openswitch, I'd like to use some of the lines for analog sets for the breakroom, kitchen, etc where they don't need all the cool features, and the other lines for POTS/DID trunks. Also how mature is this for production environment? I envision using mostly VOIP phones, cisco 7960 or Uniden UIP200 and using the voicetronix to bring in DID trunks/POTS lines. I've read reports about echo problems, is it still an issue with asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Emailing phone messages?
Where do you set the outgoing mail server for use with asterisks mail system? I have entered the info in the voicemail.conf file correctly, but I am still unable to get the voicemail messages via email. I ran a tcpdump on the system while calling in and leaving a voicemail and I don't even see the system try and contact a mail server. HELP!!! Thank you all in advance. Sean Garland Siskiyou Technology Consultants ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729 Codec+packet loss concealment
Andrew Kohlsmith [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: My qualification is having worked on the IAX2 jitter buffer, consequently having studied how audio flows from the received frames through the jitter buffer and then via ast_translate() into the codec. Hmm... having worked on the IAX2 jitter buffer, can you tell us why trunking and jitter buffers don't get along? When trunking with nufone I get ... interesting... audio if I have a jitter buffer enabled. :-) Getting back to loss concealment for a moment, it seems to me that we could do something like the following: * Every 20ms, call a scheduled function that inserts a silent voice frame into the stream. The frame would be marked as bogus in some way and would be timestamped appropriately. * The jitter buffer should then remove the duplicate voice frames, leaving a constant 20ms stream of either voice data or silence. * The individual codecs should then either spot the frame's bogus marker and deal with it as a dropped frame or, if the codec can't do reconstruction, process the frame as silent audio. I expect that a silent frame would sound much the same as a dropped frame (with no reconstruction) anyway. Does that sound feasible with the current framework? My initial inspection of the SIP/IAX2 code says that it should be, although it'd introduce a fair amount of overhead. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf
Not in configs or /etc/asterisk/. Asterisk is still running, just curious why I am not seeing that file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Jackson Sent: Tuesday, August 03, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] features.conf -Original Message- From: AJ Grinnell [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 10:28 AM To: Asterisk Subject: [Asterisk-Users] features.conf Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it? I think that it should be in configs/features.conf.sample unless you have run make samples in which case this file is copied to /etc/asterisk. Robert Jackson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned by Arialink for dangerous content and is believed to be clean. For more information please email [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Call Redirection / Transfers
I have a PRI comming into each of 2 buildings. How do I redirect an incomming call on PRI_A of particular DIDs to arrive at PRI_B instead? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hook-flash timing
Finally, can I turn off the '#' to transfer, since we're using the hook-flash (albeit manually) instead? ISTR an option to do this but have spent the morning trying to find it again unsucessfully... I think you might want to look at the 'T' and 't' options on the Dial application, documented somewhat here: http://www.voip-info.org/wiki-Asterisk+cmd+Dial jl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Emailing phone messages?
On Tue, 2004-08-03 at 11:22, Sean Garland wrote: Where do you set the outgoing mail server for use with asterisks mail system? It uses the command '/usr/sbin/sendmail -t' by default. You can use the mailcmd parameter in voicemail.conf to override that. From the wiki: Mailcmd allows the administrator to override the default mailer command with a defined command. Mailcmd takes a string value set to the desired command line to execute when a user needs to be notified of a voice mail message. http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog channel stays offhook
Hi,
We are having a problem with asterisk detecting that an analog ext has been
put down. This seems only to happen after a number of calls have been made.
We have an FXO port (TDM400P with FXO module) connected to our PBX and are
using this to test asterisk prior to rolling our for our small office.
What happens is that we make a number of calls to this ext which 1st rings
a phone (FXS) then after a timeout calls a sip phone. This seems to work
well but after a while it stops working. What we found is that the analog
ext seems to be busy ie: it has not hung up.
What we see using 'zap show channel 10' is as follows:
debian*CLI zap show channel 10
Channel: 10
File Descriptor: 16
Span: 3
Extension:
Dialing: no
Context: from-analog-incoming
Caller ID string:
Destroy: 0
Signalling Type: FXS Kewlstart
Owner: Zap/10-1
Real: Zap/10-1
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Actual Hookstate: Offhook
It seems to remain in this state until we reboot the box.
This is our zapata.conf for the context in question:
[channels]
language=en
context=from-analog-incoming
signalling=fxs_ks
usecallerid=yes
callprogress=no
busydetect=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
channel = 10-11
We have tried it with callprogress=no and callprogress=yes
This is the context from extensions.conf
[from-analog-incoming]
exten = s,1,Dial(${PHONE1}${PHONE2},10)|tTr
;exten = s,1,Dial(SIP/1337,10)|tTr
exten = s,2,Dial(SIP/dan,10)|tTr
exten = s,3,Congestion
exten = s,4,Hangup
exten = s,102,Voicemail2(b500)
exten = s,103,Congestion
exten = s,104,Hangup
exten = i,1,Hangup
exten = h,1,Hangup
Any ideas ?
Regards
Dan
PRO-NET INTERNET SERVICES LTD
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Re: [Asterisk-Users] features.conf
- Original Message - From: AJ Grinnell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 8:02 AM Subject: RE: [Asterisk-Users] features.conf Not in configs or /etc/asterisk/. Asterisk is still running, just curious why I am not seeing that file. Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's been in there for over a week now, I just checked out a new copy and it's in there... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec
Adam Hart wrote: Steve Underwood wrote: Adam Hart wrote: Daniel Niasoff wrote: Is G729 more sensitive to packet loss or delays due to its higher compression. If Ive generally got the bandwidth available, am I best sticking to ulaw. G.729 has lost packet concealment, G.711 doesn't. G.711 will sound better otherwise if you can afford the bandwidth. Eh? G.729 has no particular features to allow more effective packet loss concealment. iLBC has, but at the cost of a substantially higher bit rate. In fact G.711 is a little ahead of G.729 in the regard, since packets are completely independant. The smoothing in G.729 means you need the previous packet to decode the current one properly. Regards, Steve I believe you're mistaken - G.729 works similar to iLBC and speex. iLBC works better as the packets are independent but G.729 still has a function for packet loss concealment. prehaps have a look at http://www.speex.org/comparison.html It would probably help if you understood what that table means. It is very misleading. G.729 has features to mitigate the awfulness of a lost packet. It has nothing to help conceal lost packets really well. What I said is correct. If you fudge over a lost G.711 packet it has less bad effect than fudging over a lost G.729 packet. There is no missing smoothing data, so at least the packets you have are handled properly. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZyXEL 2000w In Call Menu/Hold configs
Hi Everyone,
After a fair amount of faffing ive managed to get the 2000w working with
asterisk for IP - PSTN calls (i.e. get the phone to make and receive calls
over our BT line). The final solution is to set up outgoing VoIP calls but
I now know that without a SIP aware router I can think again! (damn you
iptables!)
In the mean time I'm trying to figure out why I can't get the Zyxel to give
me the hold and transfer options on the screen, thus allowing me to pass
calls around once they are inside our network. I'm expecting a few desk
phones to turn up in a few weeks, but zyxel are adamant that this thing
supports it, so does anyone know how to get it working?
I was led to believe that you would need the 't' at the end of the dial
string to enable the called party to transfer the call about, is this
correct?
My dial plan for the relevant contexts looks a little like this:
[ Context 'local-extensions' created by 'pbx_config' ]
'0' =1. Goto(2000|1) [pbx_config]
'2001' = 1. Dial(SIP/2001|20|tr) [pbx_config]
2. Voicemail(u2001) [pbx_config]
102. Voicemail(b2001) [pbx_config]
103. Hangup() [pbx_config]
'2002' = 1. Dial(SIP/2002|20|tr) [pbx_config]
2. Voicemail(u2001) [pbx_config]
102. Voicemail(b2001) [pbx_config]
103. Hangup() [pbx_config]
[ Context 'always-out-pots' created by 'pbx_config' ]
'_9XX.' = 1. Dial(Zap/1/WW${EXTEN:1}|tr)[pbx_config]
2. Goto(102) [pbx_config]
102. Congestion() [pbx_config]
103. Hangup() [pbx_config]
[ Context 'from-analog' created by 'pbx_config' ]
'h' =1. Hangup() [pbx_config]
'i' =1. Hangup() [pbx_config]
's' =1. Dial(SIP/2001SIP/2002|45|tr) [pbx_config]
2. VoiceMail(u2001) [pbx_config]
3. Hangup() [pbx_config]
102. VoiceMail(b2001) [pbx_config]
103. Hangup() [pbx_config]
If anyone has any advice it would be appreciated
Regards,
jd
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RE: [Asterisk-Users] G729 Codec+packet loss concealment
If this works, wouldn't it fix the problem using silence supression as well -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Tuesday, August 03, 2004 11:22 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] G729 Codec+packet loss concealment Andrew Kohlsmith [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: My qualification is having worked on the IAX2 jitter buffer, consequently having studied how audio flows from the received frames through the jitter buffer and then via ast_translate() into the codec. Hmm... having worked on the IAX2 jitter buffer, can you tell us why trunking and jitter buffers don't get along? When trunking with nufone I get ... interesting... audio if I have a jitter buffer enabled. :-) Getting back to loss concealment for a moment, it seems to me that we could do something like the following: * Every 20ms, call a scheduled function that inserts a silent voice frame into the stream. The frame would be marked as bogus in some way and would be timestamped appropriately. * The jitter buffer should then remove the duplicate voice frames, leaving a constant 20ms stream of either voice data or silence. * The individual codecs should then either spot the frame's bogus marker and deal with it as a dropped frame or, if the codec can't do reconstruction, process the frame as silent audio. I expect that a silent frame would sound much the same as a dropped frame (with no reconstruction) anyway. Does that sound feasible with the current framework? My initial inspection of the SIP/IAX2 code says that it should be, although it'd introduce a fair amount of overhead. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP experiences with Cable and DSL
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, A thought occurred to me; Background; In the early days of cable, the cable people seemed clueless to things like over selling bandwidth. But as time went along they got it better and better under control. Of course their natural competitor, DSL, created a bigger demand for them to get things under control. Today cable is often giving 3-4Mb down and 384Kb up, and DSL is usually 768Kb down and 384Kb up. (At least in my area.) Under normal Internet use all we really care about is downspeed. So cable is providing 68.27Kb/$-91.02Kb/$ and DSL 26.48Kb/$ making cable the easy choice. But with VoIP it has to go both ways and things like latency can easily become a big issue. (I have cable and it seems that I get sound degradations much easier than I'm comfortable with, yes it's a shared connection with occational POP traffic. Also, I'm only talking about dedicated network connections for final implementation.) So, what I realized was that I have no real data to operate with is, and has anyone done an evaluation of typical needs which shows DSL better suited for VoIP? F.ex. cable shares the pipe and unless QoS is implemented can reasonably have more traffic issues than DSL. This could easily make DSL to be better suited as focus shifts to up speed. Of course DSL has a narrow maximum length tolerance and so that can also be an issue (bad implementation). For a few years now I've operated with cable as the obvious choice, at least in my area where RoadRunner really built up a good network. It could be that for nation wide implementation VoIP really should be on DSL. (Unless of course you need a big pipe where a split T is the only higher option.) - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBD7g2ljK16xgETzkRAk3KAKCYoqXV5EWAlgBMnKNo6A2CZDojEACgp7i9 q0Ldk/oCsuH8uDVOtQ/h/Tk= =zikJ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec
[EMAIL PROTECTED] wrote: On Tue, 3 Aug 2004, Steve Underwood wrote: Eh? G.729 has no particular features to allow more effective packet loss concealment. iLBC has, but at the cost of a substantially higher bit rate. In fact G.711 is a little ahead of G.729 in the regard, since packets are completely independant. The smoothing in G.729 means you need the previous packet to decode the current one properly. For IAX2, at least, Asterisk oes not use the lost-packet-concealment of any codec. This is because the incoming frames clock Asterisk. For iLBC's lost packet concealment to work, Asterisk would have to start calling the decoder with a NULL at the point when the missing packet shold have arrived. This certainly explains why we get terrible audio at 10% packet loss between Asterisk servers between 2 end points using iLBC, but if we use 2 SPA2000s using G.729 to commincate directly with each other (and having the same 10% packet loss), they sound pretty good. We had been trying to figure why iLBCs loss concealment wasn't helping much. I was never able to explain this until now:) Thanks. -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI Call Redirection / Transfers
If you want complete control from the Asterisk side you'd probably need a 2B transfer facility enabled on the PRI to allow Asterisk to tell the central office to shunt the call elsewhere. However, that functionality isn't in Asterisk yet: http://www.voip-info.org/wiki-Asterisk+bounty+PRI+2B+channel+transfer -Original Message- From: John Harragin [mailto:[EMAIL PROTECTED] Sent: August 3, 2004 8:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI Call Redirection / Transfers I have a PRI comming into each of 2 buildings. How do I redirect an incomming call on PRI_A of particular DIDs to arrive at PRI_B instead? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec
Andres wrote: [EMAIL PROTECTED] wrote: On Tue, 3 Aug 2004, Steve Underwood wrote: Eh? G.729 has no particular features to allow more effective packet loss concealment. iLBC has, but at the cost of a substantially higher bit rate. In fact G.711 is a little ahead of G.729 in the regard, since packets are completely independant. The smoothing in G.729 means you need the previous packet to decode the current one properly. For IAX2, at least, Asterisk oes not use the lost-packet-concealment of any codec. This is because the incoming frames clock Asterisk. For iLBC's lost packet concealment to work, Asterisk would have to start calling the decoder with a NULL at the point when the missing packet shold have arrived. This certainly explains why we get terrible audio at 10% packet loss between Asterisk servers between 2 end points using iLBC, but if we use 2 SPA2000s using G.729 to commincate directly with each other (and having the same 10% packet loss), they sound pretty good. We had been trying to figure why iLBCs loss concealment wasn't helping much. I was never able to explain this until now:) Thanks. If you have 10% packet loss G.729 should sound awful. Are you really getting 10% packet loss in the G.729 case? If not, why does iLBC give that? Is the higher bit rate of iLBC pushing things over your available bandwidth limit? Seems pretty odd. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rodopi Billing
Well, we currently use Rodopi and are trying to find away to use it for our VOIP billing. However, because it's based on radius I'm unsure if it will be suitable for Asterisk. I was just curious if anyone else has used the 2 together before. - Darren On Sat, 2004-07-31 at 13:47, [EMAIL PROTECTED] wrote: Rodopi is a radius system. Just build your own using freeradius. Are you using Cisco that you need radius? Cheers Clive On Fri, 30 Jul 2004 11:44:14 -0700 Darren Bentley [EMAIL PROTECTED] wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec+packet loss concealment
On Tue, 3 Aug 2004 05:47:59 -0400 Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 03 August 2004 02:58, [EMAIL PROTECTED] wrote: From what I have heard, Asterisk does not allow for iLBC to take advantage of the lost packet concealment. I understand this has something to do with the jitter processing. Can you provide a source for that statement? I am not disputing it but I'd like to have it in the archives for one, but also to verify the claim too. Regards, Andrew Hi Here I am quoting Steve Davies: For IAX2, at least, Asterisk does not use the lost-packet-concealment of any codec. This is because the incoming frames clock Asterisk. For iLBC's lost packet concealment to work, Asterisk would have to start calling the decoder with a NULL at the point when the missing packet should have arrived. Can't say for sure for SIP, but I'd guess that its the same. Steve Regards Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Herbalife Independent Distributor http://www.healthiest.co.za ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf
Chris Shaw [EMAIL PROTECTED] wrote: Not in configs or /etc/asterisk/. Asterisk is still running, just curious why I am not seeing that file. Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's been in there for over a week now, I just checked out a new copy and it's in there... Or simply rename musiconhold.conf as features.com and restart Asterisk. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rodopi Billing
On Fri, 30 Jul 2004, Darren Bentley wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? I suffered with Rodopi for three years in a previous life. Avoid it like the plague. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] Best Linux for Asterisk
On Mon, 02 Aug 2004 12:08:34 +1200, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: be it. Most of the unstable behavior has been in GUI based parts: Gnome in particular. Since no sane person runs * on a machine that is also running X, it's a non-issue. Is this always going to be the case? Is there no way of saying X doesn't get what it wants unless noone else wants it? I have honestly never seen the point of running X on a server which is running Asterisk. If you can't have your Asterisk crash (commercial/production environment) all you should have installed is Asterisk and what you need to build Asterisk and it's related tools. I know there are differing opinions on this, and I realize that yes, Asterisk does work when you install all sorts of things, but I feel the ideal, and correct thing to do with an Asterisk install is to dedicate the machine to it, run anything else you need on a seperate server. Just my .02 CDN Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf
Kevin Walsh wrote: Chris Shaw [EMAIL PROTECTED] wrote: Not in configs or /etc/asterisk/. Asterisk is still running, just curious why I am not seeing that file. Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's been in there for over a week now, I just checked out a new copy and it's in there... Or simply rename musiconhold.conf as features.com and restart Asterisk. no.. WRONG. rename parking.conf, as parking.conf is what features.conf is derived from. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec
This certainly explains why we get terrible audio at 10% packet loss between Asterisk servers between 2 end points using iLBC, but if we use 2 SPA2000s using G.729 to commincate directly with each other (and having the same 10% packet loss), they sound pretty good. We had been trying to figure why iLBCs loss concealment wasn't helping much. I was never able to explain this until now:) Thanks. If you have 10% packet loss G.729 should sound awful. Are you really getting 10% packet loss in the G.729 case? If not, why does iLBC give that? Is the higher bit rate of iLBC pushing things over your available bandwidth limit? Seems pretty odd. Bandwidth at the mentioned test server is 256MB and usage consumption was only 20-25kbps during the test. Packet loss is at the ISP (confirmed by them). I attribute the bad audio on iLBC due to Steve's explanation that Asterisk does not do any loss-concealment, but a direct call between 2 Sipuras does (no Asterisk involved). The SPA manual clearly states that The SPA applies an error concealment algorithm to alleviate the effects of packet loss. We have tested this under many conditions of packet loss and the SPA sounds great up to about 10-12% packet loss. Even at 20% loss you can carry a convesation that sounds like a mediocre cellphone call. Regards, Steve -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Tapi Driver for Windows
Hello, I am attempting to get the asttapi driver working on Windows XP Professional, and am running into some strange problems. I've combed the Web and the Wiki for information on debugging the application to see if I can solve my issue, but nothing is helping me. I have tried the driver on two different Windows XP Pro workstations with the exact same results. Here is where I grabbed the code: http://sourceforge.net/projects/asttapi/ I did the installation and verified that I could connect to the manager interface as the user that I want to use. However, whenever I try to make a call with Outlook, I am give the following error message: The selected line or address is in use by another program or device. Your call could not be placed at this time. Try your call again later. In true microsoft tradition, this is the extent of the error message, and I have found nothing related to it in any log files. I did find the following on Microsoft's Website: http://support.microsoft.com/default.aspx?scid=kb;en-us;q194253 This offers no help. Anyone have any suggestions? Anyone using the driver under Windows XP? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Rodopi Billing
On Fri, 30 Jul 2004, Darren Bentley wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? I suffered with Rodopi for three years in a previous life. Avoid it like the plague. OMG.. I had to support a rodopi installation myself for 2 years.. Closest I've ever come to suicide. While I have not managed another system but RODOPI, I have to say, there must be better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] features.conf
Josh Roberson [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Or simply rename musiconhold.conf as features.com and restart Asterisk. no.. WRONG. rename parking.conf, as parking.conf is what features.conf Oops. I knew it was one of them. At least I didn't say sip.conf :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Today's possible problems with Broadvoice????
Uhm... So ATT pays you back WHAT for the time they're done? So if they go down all-day Monday, I'll get back... A dollar? Heck, my cellular provider does better than that. That's not an SLA, that's a simple refund-agreement... Nobody makes you pay for service you don't receive. It's also completely useless, because it doesn't take into consideration damages which my arise from the missing service, nor does it punish ATT for failing to provide reliable service. The worst SLA I ever accepted called for a full day's refund if the service went down for up to 3 hours in any one 7-day period, a week's refund if the service went out for 3-6 hours, and a full month's refund for any outtage above. It would also allow penalty-free termination of the contract if ever there the service was less than 99.2% availability in any given 30-day period. Better SLAs I've signed would provide 99.5-99.9% availability and carry damage estimates for service outtages. -Original Message- From: Brian McManus [mailto:[EMAIL PROTECTED] Sent: Monday, August 02, 2004 2:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Today's possible problems with Broadvoice Absolutely, there is VoicePulse, BroadVox, Nufone, etc. Also If you want exceptional stability and don't mind paying the man ATT also has business and residential VoIP service (it's a bit spendy but very reliable, and for business a rep told me they have 100% Service Level agreements, if they go down, they will pay you back for that time of unavailable service.) : http://www.usa.att.com/callvantage/home.jsp? B Surely there are other providers to investigate, or a customer service desk at Broadvoice to complain to. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Sparc64
Ok, I may have spoken to early, I have * compiled and running on Sparc64/Linux, tried to configure sip softphones etc., everything works till here. Yesterday I tried to place a call to the demo but right after the call is bridged with the demo sounds it receives a SIGBUS and terminates with Bus error Any ideas how can I track this down? Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf
Kevin Walsh wrote: Josh Roberson [EMAIL PROTECTED] wrote: no.. WRONG. rename parking.conf, as parking.conf is what features.conf Oops. I knew it was one of them. At least I didn't say sip.conf :-) True that. This is another reminder that everyone needs to make sure that when they update, they check all of the files in the configs/ path in the src tree to see what's changed. Also, if you're confused about why something that's supposed to be in cvs isn't, a good method would be to make clean; make update; make install. If that still doesn't cure it, blow away the source tree and start with a new checkout. Just a friendly reminder to the list. twisted ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Sighting
I just got my copy of 'VON Magazine' and there is a 1 page article about Asterisk titled, SIP at RISK and Asterisk. Here is a small quote: NAT is the place where SIP messes up the worst-an IP address in the payload of a SIP signaling packet, generated on one side of a NAT, is likely to be meaningless on the other side of the NAT, after address translation. Then the article talks about Asterisk and IAX. Page 63 July/August 2004 --- [This E-mail scanned for viruses by Virus Hunter at itechgroup.com] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Integration with Altigen
I would like to integrate * with an existing Altigen PBX. I want to spend as little money as possible to make it happen. My main goal is to inexpensively connect a branch office to the phone system. Eventually I would like to replace the Altigen system with an Asterisk server so I don't want to spend any money on Altigen hardware. Currently the Altigen has analog interfaces with a couple of open ports. If I used a TDM40B (FXS ports) could I interface that with the Altigen system and connect the two of them together. Thanks, Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Rodopi Billing
Well, can anyone recommend a full featured ISP billing system that would handle VOIP/Asterisk? - Darren On Tue, 2004-08-03 at 11:09, Brian D'Arcy wrote: On Fri, 30 Jul 2004, Darren Bentley wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? I suffered with Rodopi for three years in a previous life. Avoid it like the plague. OMG.. I had to support a rodopi installation myself for 2 years.. Closest I've ever come to suicide. While I have not managed another system but RODOPI, I have to say, there must be better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI Call Redirection / Transfers
Under what circumstances? If the first T1 is down, for example? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Harragin Sent: Tuesday, August 03, 2004 7:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI Call Redirection / Transfers I have a PRI comming into each of 2 buildings. How do I redirect an incomming call on PRI_A of particular DIDs to arrive at PRI_B instead? Thanks, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and RT
Has anyone integrated asterisk with current version of rt. I followed the Wiki but I only get as far as hold on while i create a ticket then it hangs up. I don't see it connect to the rt-soap-server.pl script running on the console of my rt machine. any help would be greatly appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rodopi Billing
Darren Bentley wrote: Well, can anyone recommend a full featured ISP billing system that would handle VOIP/Asterisk? There is not one solution. Canned billing solutions never work. Write your own. If you cannot code, hire someone that can. (not me) Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rodopi Billing
While we have not integrated the asterisk CDRs yet it should not be a problem to do. We our building a billing system for ISP/CLECs that will do what you want. If you want more information you can contact via email to [EMAIL PROTECTED] or by calling 910.402.5010 Regards, Gary Carr President/CEO 705A Wesley Pines Rd. GSC Telecommunications, Inc. Lumberton, NC 28358 Phone: 910-402-5011 Fax: 910-618-9027 Check us out at: www.gsctele.com Well, can anyone recommend a full featured ISP billing system that would handle VOIP/Asterisk? - Darren On Tue, 2004-08-03 at 11:09, Brian D'Arcy wrote: On Fri, 30 Jul 2004, Darren Bentley wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? I suffered with Rodopi for three years in a previous life. Avoid it like the plague. OMG.. I had to support a rodopi installation myself for 2 years.. Closest I've ever come to suicide. While I have not managed another system but RODOPI, I have to say, there must be better. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [RANT] Today's possible problems with Broadvoice????
Rich Adamson wrote: So, for those that don't have any interest in the broadvoice interface topic, find your delete key. Its not all that hard, really. Every night when I go to bed I say my prayers. And in those prayers I include a request that those on the list who are so quick to snap at people for their cluelessness will choose instead to just delete the mail and go on. What does it cost to hit delete, and isn't it possible that a) the clueless question may be picked up by someone as a community service--which happens pretty often, or b) the niche question will actually be of intense interest to a small subset of our community, such as the Broadvoice Pioneers. Despite using spam control, I still have to hit delete fifty times or so a day to get rid of those disgusting sex ads. Why is it any harder to do the same with messages that, upon swift perusal, aren't of interest? We are asterisk's face to the new user community. Imparting clue is part of our mission, but IMO acting nice to newbies is a more important part. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Integration with Altigen
- Original Message - Message: 15 From: Geoff Nordli [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Tue, 3 Aug 2004 11:36:05 -0700 Subject: [Asterisk-Users] Integration with Altigen Reply-To: [EMAIL PROTECTED] I would like to integrate * with an existing Altigen PBX. I want to spend as little money as possible to make it happen. My main goal is to inexpensively connect a branch office to the phone system. Eventually I would like to replace the Altigen system with an Asterisk server so I don't want to spend any money on Altigen hardware. Currently the Altigen has analog interfaces with a couple of open ports. If I used a TDM40B (FXS ports) could I interface that with the Altigen system and connect the two of them together. your integration would be difficult via FX(x). the altigen is going to need to terminate the ringing line at a destination, ie vm AA that gives routing options for example. can't remember from my altigen days how well it does something like DISA but that maight work as well. going the other way would be fine, you would have to set up a dial code like 9 in the altigen that selected one of the ports connected to the *. once the port was opened you could dial anything in the dialplan in *. down and dirty but also cheap. good luck jason kawakami Open Telephony Labs, LLC www.optellabs.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog channel stays offhook
On Tue, 3 Aug 2004, Danial Subhani wrote: What we see using 'zap show channel 10' is as follows: debian*CLI zap show channel 10 Echo Cancellation: 128 taps, currently OFF Actual Hookstate: Offhook Yep - I get this too, but incoming calls still seem to arrive OK. I don't use the line for outgoing so can't comment on that. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec
On Tue, 3 Aug 2004, Steve Underwood wrote: It would probably help if you understood what that table means. It is very misleading. G.729 has features to mitigate the awfulness of a lost packet. It has nothing to help conceal lost packets really well. What I said is correct. If you fudge over a lost G.711 packet it has less bad effect than fudging over a lost G.729 packet. There is no missing smoothing data, so at least the packets you have are handled properly. In an appendix of the G711 spec there's a simple but good concealer for G711. I wanted to implement it once Asterisk can take advantage. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Play audio into meetme conference?
Is it possible to play and audio file into a meetme conference for both parties to hear? I thought I remembered reading something about it, but I can't find it now. Any help would be greatly appreciated. Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco MC3810
Wayde Nie wrote: I can get a Cisco MC3810 with a mixture of FXO and FXS ports, the MC3810 comes with a built in Ethernet port and I believe it does SIP too... Will this mean that I won't need a T1 card and dedicated channel bank? ie. Asterisk connected over Ethernet with the MC3810 and the POTS lines and stations connected to the MC3810? Does it work that way? Any other limitations or gotcha's with this approach? (I'm new to this and want to confirm before I go too far down this path...) Hi Everyone, I sent a message with the above questions over this past weekend, unfortunately I had an email service outage and don't have the thread replies to respond to in order to maintain the discussion thread... I hope this gets threaded properly ;) , apologies for the confusion if it does not... In any case Steve Szmidt responded: It's really kinda silly to have a great box like Asterisk and not use VoIP with it. Whenever you use a VoIP phone all you need is the network connection. That is the best way of using Asterisk. Maybe silly, but I have to do this with a stepped rollout approach... At first, I want to replicate what I have with POTS, except with separate extensions and other details but the user interface, aka phone handsets, remains familiar... Next, I'd like to (slowly) add the toys, IP phones, VoIP LD providers, etc... There's a good idea to have a Digium card as some Asterisk functions require a clock signal, from one of their cards. Does this mean that a digium card through the MC3810 T1 interface would provide the h/w clock whereas using Ethernet through the MC3810 10bT interface would require a less accurate s/w clock? Does anyone know if the MC3810 FXO/FXS ports are accessible through the built in Ethernet 10bT port (inferior s/w clock or not) or do you need to go in through the T1 interface? Has anyone actually done this? (I'm not really prepared to be a pioneer here ;) Grateful for any insights! Thanks, -- Wayde Nie. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom 200 - custom melody
Has anyone used this feature successfully? I 'think' I have a .wav file that it wants. Here is what 'file' says: sf-george.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz I see the logs on my web server as it tries to access it, but all I get is a screech out of the phone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco ubr924
Well one of the _big_ problems I see right now, is that the cisco ubr924 is reporting it's MGCP version as 0.1 and asterisk errors with incompatible version. Not sure if that is a cisco bug and really should be 1.0 I will upgrade the IOS and see. Maybe they do run version 0.1 but I've never seen an RFC for MGCP version 0.1 Duane Cox - Original Message - From: Duane Cox To: Gabriel Millerd Cc: [EMAIL PROTECTED] Sent: Friday, July 30, 2004 10:58 AM Subject: Re: [Asterisk-Users] cisco ubr924 I belive the 924 will do either MGCP or H323. I will start working on it today. Even if it DOES do MGCP, it's not guarenteed that * will like it. Any of your previous work (config files) would be of assistance. I am going to start on it today. Thanks Duane Cox - Original Message - From: Gabriel Millerd To: [EMAIL PROTECTED] Sent: Friday, July 30, 2004 10:29 AM Subject: Re: [Asterisk-Users] cisco ubr924 Hey list, Does anyone have a current working config example of a cisco ubr924 and * ? I think the 924 only supports MGCP. I want to get VoIP on this device, I was wondering if anyone has already tackled the problem, if not, I'll go in blind :) i have tried myself (and posted) with not avail. If its possible to learn from your progress please lemme know. If there is anything I can help with also let me know. I believe only H323 is an option for connectivity. I was never successful in getting it to work with opengk and the like. However its not like that stuff is anywhere as friendly as * is Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] instable Modem-Module in CVS ?
Hallo everyone, at first I would like to say hello to anyone as I am new to this list and new to asterisk which I find very fascinating. I am currently using asterisk with the German SIP-Provider sipgate and with my little ISDN-Line using the Modem-Driver vor I4L. I upgraded my source tree from CVS some hours ago because the stable version does not hangup SIP-Calls correctly. Unfortunately now I am confronted with nearly random crashes stating Floating Point Exceptions and many error messages channel.c:1650 ast_set_write_format: Unable to find a path from UNKN to SLINR while processing calls. The first especially appears when using the VoiceMailMain-Application while the latter appears more often (and I cannot say in conjuction with what). It also seems that using the Statement Playtones(Dial) confuses the I4L-Modem: When Dialing *8 (my command for VoiceMailMain-Application), there is a bad noise for 1/2 a second and afterwards you can hear rest of the normal Mailbox ? prompt. I hope I did not ask a stupid question and would appreciate your answers :-) Christoph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play audio into meetme conference?
Hi Il mar, 2004-08-03 alle 22:09, Paul Egger ha scritto: Is it possible to play and audio file into a meetme conference for both parties to hear? I thought I remembered reading something about it, but I can't find it now. Any help would be greatly appreciated. sure. use the call spooling file to connect to the meet me room and play any file with playback,background,mp3player,blah matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] After RC1 upgrade, temporary loss of voice
I just upgraded to RC1 from a two-three month old CVS , and noticed that during IAX2 calls to my service provider there are periods (usually less than 10 seconds long, minutes apart) during which the caller can not hear me, but I can hear the caller fine. Inter-office calls (SIP-to-SIP) does not appear to have this issue. Has any other users experienced this? Marcus Adolfsson TreoCentral Store http://store.treocentral.com/ Treo Smart Phones, Accessories, and Software Toll Free (800) 557 6819 ext 111 Direct (212) 202-8350 Fax (212) 202-8348 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?
My SPA-3000 finally arrived and I'm trying to get the FXO port on it to work as if it was a X100P card as far as Asterisk is concerned. I have Asterisk dialing out over the SPA-3000 FXO port no problem. The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk mainmenu context (or ext I guess). Currently the SPA-3000 answers the call, then I hear a modified dial tone, which if I dial any extension + #, it will ring a SIP phone no problem. So now I just need to get it to that automatically. The SPA-3000 User guide shows how to have it automatically forward incoming PSTN calls to its FXS port, but that would normally be a phone, not Asterisk. Any ideas? -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] After RC1 upgrade, temporary loss of voice
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 03 August 2004 05:14 pm, [EMAIL PROTECTED] wrote: I just upgraded to RC1 from a two-three month old CVS , and noticed that during IAX2 calls to my service provider there are periods (usually less than 10 seconds long, minutes apart) during which the caller can not hear me, but I can hear the caller fine. Inter-office calls (SIP-to-SIP) does not appear to have this issue. Has any other users experienced this? Marcus Adolfsson TreoCentral Store http://store.treocentral.com/ Treo Smart Phones, Accessories, and Software Toll Free (800) 557 6819 ext 111 Direct (212) 202-8350 Fax (212) 202-8348 Hmm, I've been experiencing the same today. Though I'm not on RC1, I'm on HEAD-07/07, and I just recently implemented g729. Which is more sensitive to dropped packets. Otherwise I've seen this occur when one party have network traffic on his side. The other party cannot hear them but the one with busy connection could. Guess it depends on in what direction the heavy traffic is going. I had no traffic on my side, though. Also, I'm going to turn on QoS on my router, to prioritizing VoIP, that might make all the difference since RR does support it. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBEAUNljK16xgETzkRAgLYAKCeOUSKS0FhuAnOOKrQV9ZzZ1XVFACgmJEH w7PMdgMKSxyFfrB4rkK6k3k= =chqf -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?
The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk mainmenu context (or ext I guess). Configure an auto-dial number in the SPA to that it corresponds to something in the mainmenu context. Like: PSTN_Caller_Default_DP[2] 2 ; Dial_Plan_2[2](S0:551155) ; When a call comes in the FXO port, the SPA automatically dials 551155 via your Proxy[2] settings.. -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 as a regular Asterisk FXO device?
Works like a charm Andres. Much appreciated. On Tue, 2004-08-03 at 16:44 -0500, Andres wrote: The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk mainmenu context (or ext I guess). Configure an auto-dial number in the SPA to that it corresponds to something in the mainmenu context. Like: PSTN_Caller_Default_DP[2] 2 ; Dial_Plan_2[2](S0:551155) ; When a call comes in the FXO port, the SPA automatically dials 551155 via your Proxy[2] settings.. -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can Zap detect line is already off-hook?
I have the need for a slightly odd * configuration for testing purposes. I have a working * setup with SIP softphones, VoIP trunks and a single X100P clone for PSTN access. The PSTN line I'm using for testing is also in use by other folks. For incoming calls, I'd like to set is up so that * functions as a voicemail backstop on this line. This much is working fine. For outgoing, I'd like * to be able to tell if the line is already in use by someone else - ie when it tries to take the line off hook, can it detect that the line is already off hook and return to the dialplan that the line is unavailable? -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk call parking + SNOM lighted buttons?
Dr. Chudobiak, I do not believe it is possible (yet). I know it is implemented in the snom 4s product but I am pretty sure asterisk cannot handle line appearances. I will do some further research. Please use my personal email address [EMAIL PROTECTED] since I get all the list emails to this address sometimes yours get lost. Thanks, Steve - Original Message - From: Dr. Michael J. Chudobiak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 02, 2004 4:23 PM Subject: [Asterisk-Users] asterisk call parking + SNOM lighted buttons? I'm trying to get call parking working with the lighted buttons on the SNOM 200. I have set the 5 buttons to Park Orbit, for extensions 700-704. Pressing the first button (x700) does park the call. However, the remaining buttons (x701-704) don't allow me to pick up parked calls, or show parking status via the LEDs. I can only pick up parked calling by manually dialing the 701-704 extensions. Has anyone successfully implemented a call parking scheme with the SNOM 200, which uses the lighted buttons to show the parking status? I'd liked to see at a glance which parking spots are active. I am using the SNOM 3.35 firmware, and CVS-HEAD-08/02/04-14:14:04, with the chan_sip2 module (chan_sip2A4.c). - Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK VoIP-PSTN gateway recommendations
I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with'#' transfer after hold...
Hi.. Has anybody been experiencing any problems with transfers using # to transfer after taking a call off of hold? Transfers using the # and music on hold work fine by themselves. However, when we place somebody on hold we can no longer use the # to transfer. This is a problem since we use the # button to park calls. So, say a call comes in, the operator is on a call already, places call on hold and answers the new call, places new call on hold, resumes old call and tries to transfer using the # button it wont work, itll just play the DTMF tone for the # button. At first, I thought somewhere along the line the Tt options must be messed up in a dial command somewhere.. but I double checked everywhere and ensured that I was enabling transfers. Does anybody have any suggestions? Thanks, Steve
Re: [Asterisk-Users] problems with'#' transfer after hold...
- Original Message - From: Stephen Hon To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 3:48 PM Subject: [Asterisk-Users] problems with'#' transfer after hold... Hi.. Has anybody been experiencing any problems with transfers using # to transfer after taking a call off of hold? Transfers using the # and music on hold work fine by themselves. However, when we place somebody on hold we can no longer use the # to transfer. This is a problem since we use the # button to park calls. So, say a call comes in, the operator is on a call already, places call on hold and answers the new call, places new call on hold, resumes old call and tries to transfer using the # button. it won't work, it'll just play the DTMF tone for the # button. At first, I thought somewhere along the line the 'Tt' options must be messed up in a dial command somewhere.. but I double checked everywhere and ensured that I was enabling transfers. Does anybody have any suggestions? Thanks, Steve Are you using the double ## transfer patch or just the regular single # that comes with CVS? -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP experiences with Cable and DSL
On Tuesday 03 August 2004 12:07, Steve Szmidt wrote: But with VoIP it has to go both ways and things like latency can easily become a big issue. (I have cable and it seems that I get sound degradations much easier than I'm comfortable with, yes it's a shared connection with occational POP traffic. Also, I'm only talking about dedicated network connections for final implementation.) As the old Rogers Cable and Bell HSE commercials used to slog it out with With cable you're all sharing a link, with HSE it's individual links -- there is some truth in that. You have a dedicated TX/RX interface with DSL; once you hit the DSLAM you are, of course, just part of some gigantic ATM flood but at least the bandwidth on that ATM network is likely far beyond what is normally available. With cable you're fighting to talk; something that QoS isn't going to help with in a CSMA/CD network. So, what I realized was that I have no real data to operate with is, and has anyone done an evaluation of typical needs which shows DSL better suited for VoIP? F.ex. cable shares the pipe and unless QoS is implemented can reasonably have more traffic issues than DSL. QoS isn't going to help you get to talk in a crowded CSMA/CD network. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK VoIP-PSTN gateway recommendations
Hi David- You may want to post this in the asterisk-biz section, you'll probably get more leads there.. Regards, Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gurr Sent: Tuesday, August 03, 2004 3:19 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] UK VoIP-PSTN gateway recommendations I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP experiences with Cable and DSL
- Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 4:05 PM Subject: Re: [Asterisk-Users] VoIP experiences with Cable and DSL On Tuesday 03 August 2004 12:07, Steve Szmidt wrote: But with VoIP it has to go both ways and things like latency can easily become a big issue. (I have cable and it seems that I get sound degradations much easier than I'm comfortable with, yes it's a shared connection with occational POP traffic. Also, I'm only talking about dedicated network connections for final implementation.) As the old Rogers Cable and Bell HSE commercials used to slog it out with With cable you're all sharing a link, with HSE it's individual links -- there is some truth in that. You have a dedicated TX/RX interface with DSL; once you hit the DSLAM you are, of course, just part of some gigantic ATM flood but at least the bandwidth on that ATM network is likely far beyond what is normally available. With cable you're fighting to talk; something that QoS isn't going to help with in a CSMA/CD network. So, what I realized was that I have no real data to operate with is, and has anyone done an evaluation of typical needs which shows DSL better suited for VoIP? F.ex. cable shares the pipe and unless QoS is implemented can reasonably have more traffic issues than DSL. QoS isn't going to help you get to talk in a crowded CSMA/CD network. -A. Being a cable user, the other thing I notice is that cable (or at the very least my ISP) also seems to suffer from ARP flooding... Billions and Billions of Are you there? Yes I am! Who Is at blah? I am at Blah! Crap every second, probably wasting like 512kbit of bandwidth just for DHCP and BOOTP crap... But for the most part I gotta say that the sustained transfer rates are WAY better than they ever were with DSL... And I don't notice too much difference in latency between the two... As the old Rogers Cable and Bell HSE commercials used to slog it out with With cable you're all sharing a link, with HSE it's individual links -- there is some truth in that. You guys probably remember the old ethernets where the ether was this long thick yellow cable (ThickNet) HFC is something like that, everyone is sharing the same link like with the old ThickNet and BNC networks, it is not switched at all until you get to the headend and as more people use the link, the more congested it becomes until it becomes unusable because even ARP messages can't go through... QoS isn't going to help you get to talk in a crowded CSMA/CD network. I might be misunderstanding you about QoS, but I know for a fact that it does help greatly because whether you use DSL or Cable, your bridge device (it's not a modem no matter how much people want to call it that, it's a bridge!) uses large buffered queues to achieve sustained transfer rates... this is awesome for bulk downloads but makes your VoIP conversation sound like you're on a cellphone under a bridge in a windstorm... Also if the ISP is using QoS and they classify users by the MAC address of your bridge device, they can create something similar to ATM PVCs, allowing traffic to flow more orderly and evenly across THEIR network... Bear in mind that when you're using QoS you're shaping YOUR traffic as it goes out YOUR link... you can do nothing about what happens to it once it crosses your ISP's router into the rest of the InterNet. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Rodopi Billing
Thanks for the vote of confidence guys. We just bought an ISP that uses rodopi exclusively for Accounting and Billing. ...sigh... -e -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 12:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Rodopi Billing On Fri, 30 Jul 2004, Darren Bentley wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? I suffered with Rodopi for three years in a previous life. Avoid it like the plague. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rodopi Billing
That sigh will turn to cursing after a couple of months. We currently use Rodopi, have for 10 years but the inflexability is too much to deal with anymore so we are moving away from it. Gary - Original Message - From: Ejay Hire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 7:54 PM Subject: RE: [Asterisk-Users] Rodopi Billing Thanks for the vote of confidence guys. We just bought an ISP that uses rodopi exclusively for Accounting and Billing. ...sigh... -e -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, August 03, 2004 12:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Rodopi Billing On Fri, 30 Jul 2004, Darren Bentley wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? I suffered with Rodopi for three years in a previous life. Avoid it like the plague. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
