[Asterisk-Users] Automatic setup of calls between two external lines
Is it possible to automatically set up a call between two external lines? I would like Asterisk is call a cellphone number, wait for it to answer, and then call another cellphone, when that answers connect the two together. I assume it is possible but can someone point me how to do it. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IAX2 attempts native bridge when notransfer=yes
This comment comes up fairly regularly and is confusing people. Why doesn't it say that it failed so we know? The way it is now it kind of leaves you hanging there and you don't know if the transfer happened or not. And why was it even attempted if it is obvious that transfer is off? (I know it can depend on the remote side.) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 22 July 2005 23:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX2 attempts native bridge when notransfer=yes On Wed, 20 Jul 2005, Peter Hsu wrote: Asterisk keeps attempting to do these native transfers.. Any ideas what I'm doing wrong? This is driving me crazy. Don't worry. The attempt a native transfer function is called, logs that it attempts it, then notices the notransfer=yes and fails the transfer. So nothing happens. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX over HTTP
For work environments where you only get HTTP or HTTPS access, what is the feasibility of doing IAX over HTTP? I have heard of projects such as stunnel. Has anyone tried something like this? I did a quick search but didn't come up with much. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: IAX over HTTP
Doesn't Skype use something similar? I have heard that it is encrypted and works through http proxies. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Stewart Sent: 21 July 2005 22:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: IAX over HTTP HTTP uses TCP. Too much overhead. Add SSL on to that and you have a huge amount of overhead. The end result would be poor and choppy sound quality. Jason On 21/07/05 21:58 +0200, Rob Scott wrote: For work environments where you only get HTTP or HTTPS access, what is the feasibility of doing IAX over HTTP? I have heard of projects such as stunnel. Has anyone tried something like this? I did a quick search but didn't come up with much. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Problem with DTFM and complex international setup
I don't think so. Your problem seems to do with your not being able to use an IAX client to transmit DTMF tones properly somehow. I am using a normal phone to connected to FWD which then connects to an Asteriskserver using IAX protocol. The point is that between the phone and the far Asterisk server, I guess that the tones are being sent as audio and not as inbound messages. So the far Asterisk server has to listen to the audio for the tones. On an unstable connection, it is sometimes missing a tone, or hearing a break in a tone and thinking that it is two identical tones, which results in a mis-interpreted number sequence at the other end. So I think the problems are different. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark EdwardsSent: 03 July 2005 02:20To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Problem with DTFM and complex international setup Do you think this might have an impact on http://bugs.digium.com/view.php?id=4631? Mark On 7/3/05, Mohit Muthanna [EMAIL PROTECTED] wrote: Right... that's the one. My mistake.On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote: I don't find this option in the Makefile. I find RADIO_RELAX which is something to do with radios and DTMF. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Mohit Muthanna Sent: 01 July 2005 23:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem with DTFM and complex international setup Try compiling Asterisk with RELAX_DTMF (See Makefile). Mohit. On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote: We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed Asterisk so that it is possible, using a PIN code, to dial out from Asterisk onto the local PSTN. This also works occasionally. Looking at the message from the Asterisk box it is clear that sometimes numbers are missed or repeated in the dial string. This I suspect is because Asterisk is listening to the DTMF tones but the signal is dropped; sometimes the drop means that a whole digit is dropped and sometimes is means that a digit is repeated. Does anyone know how I can fix this to make it more reliable (out-of-band DTMF?) or a better way to achieve a reliable setup? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of people. Those who understand binary, and those who don't." ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users--Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of people. Those who understand binary, and thosewho don't."___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- regards,Mark P. EdwardsFWD: 667917 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with DTFM and complex international setup
We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed Asterisk so that it is possible, using a PIN code, to dial out from Asterisk onto the local PSTN. This also works occasionally. Looking at the message from the Asterisk box it is clear that sometimes numbers are missed or repeated in the dial string. This I suspect is because Asterisk is listening to the DTMF tones but the signal is dropped; sometimes the drop means that a whole digit is dropped and sometimes is means that a digit is repeated. Does anyone know how I can fix this to make it more reliable (out-of-band DTMF?) or a better way to achieve a reliable setup? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Problem with DTFM and complex international setup
I don't find this option in the Makefile. I find RADIO_RELAX which is something to do with radios and DTMF. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohit Muthanna Sent: 01 July 2005 23:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem with DTFM and complex international setup Try compiling Asterisk with RELAX_DTMF (See Makefile). Mohit. On 7/1/05, Rob Scott [EMAIL PROTECTED] wrote: We have some guys working in the US who can't always dial back to our company in Europe easily (lots of clients require authorization to make international calls), so I set up the following: - ipkall.com number links to a FWD number - office Asterisk box registers with FWD Then I programmed Asterisk to accept office extension number using DTFM tones. This works OK. Then I programmed Asterisk so that it is possible, using a PIN code, to dial out from Asterisk onto the local PSTN. This also works occasionally. Looking at the message from the Asterisk box it is clear that sometimes numbers are missed or repeated in the dial string. This I suspect is because Asterisk is listening to the DTMF tones but the signal is dropped; sometimes the drop means that a whole digit is dropped and sometimes is means that a digit is repeated. Does anyone know how I can fix this to make it more reliable (out-of-band DTMF?) or a better way to achieve a reliable setup? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] There are 10 types of people. Those who understand binary, and those who don't. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] How to Configure a H323 Phone (newbie here)
I would also be interested. I've tried several times unsuccessfully to set up H323 with Asterisk. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adeel -31Sent: 01 July 2005 23:32To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] How to Configure a H323 Phone (newbie here) i read that asterisk supports iax,sip and h323 protocols i've used sip iax softphones ... now i've a hardphone... an IP phone (Netphone) that supports h323 . i've compiled pwlib ,oh323 and asterisk -oh323 successfully ... but i m unable toplace calls to/bymy phone... i m confused whether to use h323.conf or oh323.conf and how ? i think it's different from iax.conf sip.conf can anyone send me his working oh323.conf . or give some link that can be helpful in configuration Adeel Yahoo! SportsRekindle the Rivalries. Sign up for Fantasy Football ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Debugging zaphfc + PBX integration
If you your board into an ISDN wall socket and it works then you are acting as a terminal so you are in terminal mode. Now, how are you connecting to the PBX? If you are connecting to an ISDN extension on the PBX, then still you have to match the kind of connection, whether it is point-2-point or point-2-multipoint. Then it should work. If you are connecting it to an external line on the PBX, then it will have to be in station mode and also you have to work out p2p or p2mp and also have an ISDN crossover cable (not an ethernet crossover cable). So a lot for you to work out. I have successfully got a zpahfc card working on an extension of our PBX and also a PRI card working on an external PRI line of the PBX, so it should work for you in the end. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] cannot dial two phones using zap
Looks normal to me. What Dial with the '' means is that both lines ring, but the first one to answer is connected on the call. From you trace it looks like Zap/3-1 which is your number 206 answered the call, so the other line goes to hangup. The Dial with '' is used to implement call teams where all the phones ring but the first one to pickup gets the call and the other phones stop ringing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eddie Sent: 14 April 2005 05:53 To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] cannot dial two phones using zap I cannot dial two phones using zap at the same time. One will ring but the other one hangs up. zapata.conf [channels] context=default signalling=fxs_ks immediate=no busydetect=yes callprogress=no echocancel=yes echocancelwhenbridged=yes usecallerid=yes usecallingpres=yes threewaycalling=yes transfer=yes callerid=Incoming 20941261 group=1 channel = 3,4 extensions.conf [internal] exten = 300,1,Dial(Zap/3/206Zap/4/221,15) exten = 300,2,Hangup CLI linux*CLI dial [EMAIL PROTECTED] -- Executing Dial(OSS/dsp, Zap/3/206Zap/4/221|15) in new stack -- Called 3/206 -- Called 4/221 -- Zap/3-1 answered OSS/dsp -- Hungup 'Zap/4-1' Console call has been answered Please advice. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Acceptable voice time delay
Around 250ms max. Over that and you will have the walkie-talkie effect you are experiencing. So with you 600ms delay you are way over the top. There is also the delay on the call on the PSTN side you have to take into account. For example, I am in Europe and making a call to the UK via Voipjet is usually OK. But making a call to Romania is a lottery. Sometimes it is great, sometimes the delay is huge. And that has nothing to do with the internet delay which is more or less constant at around 180ms for me. But with 600ms, which is over half a second, you have problems. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Zap - What is going on?
For a start it should be
${EXTEN}
You have to realize that ALL variables look like that.
Dollar-open-curly-brackets-variablename-close-curly-brackets.
So it didn't see your text as a variable and it tried to call the number
$EXTEN on Zap/g2.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Etienne
Pretorius
Sent: 04 April 2005 13:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Zap - What is going on?
Ok - I was told that you set a group for Zap channels.
So I tried to make use of my Zap channels so the 2 I am interisted
in is channel 3 and channel 4.
I make Channel 3 in use bu calling a line... then I try to call another
line so expecting to have Zap channel 4 open and allowing me to make a
call, but it just keeps on ringing... and then times out. Can anyone
please shed some light on this for me?
extensions.conf
[outgoing] ;Dial
0 on the phone for external line
;SIP
Phones need another way... they act like a cell phone
exten = _0,1,Dial(Zap/g2/$EXTEN,20,tr) ;Try
finding a line...
exten = _0,2,Goto(_0-${DIALSTATUS},1) ;Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = _0-ANSWER,1,Goto(_0,102)
exten = _0-.,1,Goto(_0,1) ;Try
another line
exten = _0,102,Congestion
exten = _0,103,Hangup
Asterisk Console:
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/g2/$EXTEN|20|tr) in new stack
-- Called g2/$EXTEN
-- Nobody picked up in 2 ms
-- Hungup 'Zap/4-1'
===That channel is free and
has a seperate phone line connected to it.
-- Executing Goto(Zap/1-1, _0-NOANSWER|1) in new stack
--
Kind Regards
Etienne
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[Asterisk-Users] LDAP and Asterisk
I am looking to roll out an Asterisk VoIP implementation to our 200 employees. So far I have hooked up the Asterisk box to our Elmeg PBX via a PRI interface card and have that working, plus about 30 test users on Xlite softphones. Up til now all the configuration has been done by hand editing extensions.conf and sip.conf and voicemail.conf as needed. I would rather this was kind of automatic - when a new user is created then everything is already setup for them. We are in a (horror of horrors) Microsoft environment running Windows XP, Windows 2003 Server with AD and a sizable number of Sun and Linux boxes for development (we are an IT development shop). So what springs to mind is someone how connecting Asterisk to AD and using some spare fields in AD to hold extension numbers and the like and querying through an LDAP interface. Kind of like Realtime but using LDAP. Does anything like this currently exist? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Does IAX supports silence suppression?
Short answer is no. You should always turn it off on any client you have. Longer answer is that is is being worked on and should be available any day now (although that has been the case for some months). Also someone is working on porting it to SIP as well as IAX2. No idea if the new work will tell your if the client is using silence suppression. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcin Okraszewski Sent: 25 March 2005 12:14 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Does IAX supports silence suppression? Hi, Does IAX supports silence suppression? If yes, is there any way to detect that the other party has turned on silence suppression and there is no packet loss? Is (Halt|Reasume) audio/video transmission control messages used for this reason? Regards, Marcin Okraszewski ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Zaphfc + PRI card problem
I will reply to my own question in a case the answer is of use to others. I rebooted my Asterisk server and now it seems to work OK again. I suspect that after a while the zaphfc BRI interface has problems and needs a reboot. I have seen different problems like this, i.e. problems that occur over time especially with answering calls, with the zaphfc drivers before. A reboot usually cures it for a while, but it would be great if such a thing wasn't needed. -Original Message- From: [EMAIL PROTECTED] on behalf of Rob Scott Sent: Wed 3/23/2005 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Zaphfc + PRI card problem I have the latest bristuff, a zaphfc card for external calls and a PRI card for connecting to a PBX as a channel bank. With a BRI I would expect to be able to have two incoming calls going at the same time, but when I try it, one call connects and the other gives the following console message, a busy tone, and then a hangup: Mar 23 19:18:35 WARNING[5595]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! Mar 23 19:18:37 WARNING[5595]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! -- Channel 0/2, span 1 got hangup Anyone know what this means? I guess I should be able to have two simultaneous incoming calls over a BRI card, no? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaphfc + PRI card problem
I have the latest bristuff, a zaphfc card for external calls and a PRI card for connecting to a PBX as a channel bank. With a BRI I would expect to be able to have two incoming calls going at the same time, but when I try it, one call connects and the other gives the following console message, a busy tone, and then a hangup: Mar 23 19:18:35 WARNING[5595]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! Mar 23 19:18:37 WARNING[5595]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! -- Channel 0/2, span 1 got hangup Anyone know what this means? I guess I should be able to have two simultaneous incoming calls over a BRI card, no? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Voice getting cutoff
Looking at that list, the easiest way would be to disable all your USB ports in your BIOS, reboot and see if the card has its own IRQ. Assuming you don't need USB. In general, just turn off all the things you don't need that use IRQs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Saturday, March 19, 2005 5:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Voice getting cutoff How can I change the IRQ of the cards? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Viernes, 18 de Marzo de 2005 12:15 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voice getting cutoff Anton Krall wrote: What do you think? CPU0 0: 16148159 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 0 XT-PIC usb-uhci 8: 1 XT-PIC rtc 10: 161351663 XT-PIC usb-uhci, wcfxo 11:1276097 XT-PIC usb-uhci, eth0 12: 161350551 XT-PIC ehci-hcd, PS/2 Mouse, wcfxo 14: 138574 XT-PIC ide0 15: 33 XT-PIC ide1 NMI: 0 ERR: 0 Any problems here? Yes. Digium cards must be on their own IRQ or you will have weird problems. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Goto and E1 line
You should have set up the two cards as zaptel as a different group in
the zapata.conf.
Then if you want to dial your pbx you are dialing out of Asterisk, so
you use the Dial command.
Assuming that the PBX PRI link is in group 2 in zapata.conf
Something like:
exten = ,1,Dial(Zap/g2/${EXTEN})
Is what you want if you are dialing extension on your hard PBX.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
SALMON
Sent: Saturday, March 19, 2005 1:16 PM
To: Liste User Asterisk
Subject: [Asterisk-Users] Goto and E1 line
Hi,
I have a server with 2 TE110P cards. 1 card is plugged to telco line,
another card is plugged with a Hicom PBX.
I want to send some call to VoIP phones and all other to my PBX.
I don't known how to make my dialplan :
===Extensions.conf==
[incoming_call]
exten = 09020,1,Goto(callcenter,100,1)
exten = 022956353,1,Goto(callcenter,100,1)
exten = 022956388,1,Goto(callcenter,100,1)
exten = 022956355,1,Goto(callcenter,101,1)
exten = s,1,Goto(go_to_pbx,200,1)
[callcenter]
exten = 100,1,Answer
exten = 100,2,SetMusicOnHold(default)
exten = 100,3,DigitTimeout,5
exten = 100,4,SetVar(QUEUE_PRIO=5)
exten = 100,5,Background(welcome)
exten = 100,6,Queue(hotline) ;VoIP Phones
exten = 101,1,Answer
exten = 101,2,SetMusicOnHold(default)
exten = 101,3,DigitTimeout,5
exten = 101,4,SetVar(QUEUE_PRIO=10)
exten = 101,5,Background(welcome_privilege)
exten = 101,6,Queue(hotline) ; VoIP Phones
[go_to_pbx]
HERE I DON'T KNOWN HOW TO DIAL MY HICOM PBX :( AND SEND THE GOOD DIALED
NUMBER
=
Someone can help me please?
Thanks
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[Asterisk-Users] Any 24 (or 30) way FXS PCI cards?
It seems to me silly to have a T1/E1 card to connect to a channel bank when you could just have a 24/30 way FXS card in the slot in the first place. Does such a thing exist? Wouldn't Digium have a lot of customers if they could produce one for say $1000 retail? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rhino channel banks
Anyone have any experience with the Rhino T1 channel bank? It looks very cost effective at around $1300 for 24 lines but I haven't seen it mentioned on the Asterisk list yet. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] TE110P experiance
I have noticed the following: - the PCI ID of the card seems to change over time which means that loading the module does not always recognise the card, only way to reset this is to power cycle the machine - you cannot unload the module once it is loaded, it hangs the machine, which also means if you have automatic shutdown scripts for restarting the machine then the machine will hang on reboot One of the LEDs shows the status of the connection. If it is off, then it is not active, i.e. zaptel drivers not loaded. If red then bad connection i.e. it is not talked to the other end, usually a wiring problem If green then everything OK. Could also be a yellow state but I haven't seen that. Once you get it working, leave the thing up is my only advice. It is a shame that it is not bug free, neither the hardware nor the software so far. I don't know what Digium want to do about the hardware. I hope there is a firmware fix rather than having to mess with the actual physical hardware. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 provides ring tone?
I have an Asterisk box with TE110P PRI connected in net mode to a PBX. Both are PRI EuroISDN. The connection seems to work OK but when calling from Asterisk to the PBX through an Xten, the Xten client does not get a ringing tone when the PBX phone rings. Is it possible to set this up? Is there some zaptel or zapata setting for this or is it something that needs to be in extensions? Thanks for you help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why Asterisk can't cope with silence suppression?
OK I have to ask. Why is it that Asterisk can't cope with silence suppression? All the clients seem to be able to but not Asterisk. What would be needed to get it to work with silence suppression? What is the problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc buffer underflow/overflow messages
I get a ton of these messages, a pair every 4 or 5 mins. Is it a problem? I am wondering where they come from and if they are important. I have a zaphfc card running in TE mode connected to a PBX. Feb 16 20:23:04 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:23:04 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 49, 49 Feb 16 20:24:09 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:24:09 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 46, 46 Feb 16 20:25:13 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:25:13 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 51, 51 Feb 16 20:26:18 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:26:18 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 48, 48 Feb 16 20:27:23 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:27:23 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 45, 45 Feb 16 20:28:27 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:28:27 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 50, 50 Feb 16 20:29:32 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:29:32 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 47, 47 Feb 16 20:30:36 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:30:36 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 52, 52 Feb 16 20:31:41 epbw202 kernel: zaphfc: b channel buffer underrun on card 0 Feb 16 20:31:41 epbw202 kernel: zaphfc: b channel buffer overflow on card 0: 49, 49 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using zaphfc and wcte11xp at the same time problem
I am having problems loading the zaphfc from bristuff and wcte11xp drivers at the same time. If I load zaphfc then all works fine. If I then load wcte11xp, the card using the zaphfc doesn't pick up calls anymore. I am using bristuff 0.2.0-RC5. Anyone else seen this problem, know of a fix, or can tell me what I am doing wrong? Thanks My zaptel.conf: # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,0,0,ccs,hdb3,crc4 bchan=4-18 dchan=19 bchan=20-34 My zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [channels] rxgain = 5.0 txgain = 5.0 switchtype = euroisdn pridialplan=local prilocaldialplan=local usecallerid=yes echocancel=yes immediate=yes ; p2mp TE mode signalling = bri_cpe_ptmp group = 1 Context = fromzap channel = 1-2 signalling = pri_net group = 2 context = fromzap channel = 4-18,20-34 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] X-Lite Softphone
Turn of Silence Supression. If you have already done that then I think you are having the usual Xlite - Asterisk experience. At least I have the same problems with it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC-S and TE110P at the same time
I guess it is possible to have an HFC-S card and a Digium TE110P card working at the same time? The TE110P will work in E1 mode. I think the zaptel.conf is probably right but the zapata.conf not (I just tacked on another group at the end but I don't really know what I am doing). Can anyone help? - My zaptel.conf looks like this: # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,0,ccs,hdb3,crc4 bchan=4-18 dchan=19 bchan=20-34 And my zapata.conf looks like this: ; ; Zapata telephony interface ; ; Configuration file [channels] rxgain = 5.0 txgain = 5.0 ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode ;signalling = bri_net pridialplan=local prilocaldialplan=local ; trust user provided callerid (clip no screening)? ;pritrustusercid = yes ;callerid=asreceived usecallerid=yes echocancel=yes ;echocancel=no immediate=yes group = 1 context=fromzap channel = 1-2 group = 2 context = fromzap channel = 4-18 channel = 20-34 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] jitterbuffers - suggested settings
Any idea when that is likely to be ready? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of joachim Sent: Tuesday, February 08, 2005 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] jitterbuffers - suggested settings I recommend to deactivate the current jitter buffer and wait till a new one is ready. Joachim. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] jitterbuffers - suggested settings
Apparantly the new one will do things like interpolation so that if packets are lost it will generate new ones to fill the gap. The current jitterbuffer doesn't do that so you get silence on packet loss. There are a bunch of other features too that I don't remember, but that was the most interesting to me last time I looked. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Schulte Sent: 08 February 2005 13:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] jitterbuffers - suggested settings ? What's wrong with the current jitterbuffer.. -Original Message- From: joachim [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 08, 2005 2:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] jitterbuffers - suggested settings I recommend to deactivate the current jitter buffer and wait till a new one is ready. Joachim. Stuart Elvish wrote: Hi, I was wondering if anyone else has a similar setup and can suggest settings for the jitterbuffer: I have a client with an ADSL connection at site A B with site A being dedicated to voice and having no Asterisk server, site B combining voice and data with traffic shaping and housing an Asterisk server. There seems to be packet loss / jitter on this connection and I wanted to know if anybody could suggest the number to put in jitterbuffers= and whether or not they have found this to affect the echo. Any suggestions will be greatly appreciated. Kind Regards Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] zaphfc
I am also interested in sound quality with respect to the zaphfc drivers. What is your physical setup? Where are you listening for the noise? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Corvin Sent: Monday, February 07, 2005 7:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] zaphfc Hi, I have strange sound artifacts when someone calls me, even if nobody is speaking tx bar in ztmonitor is moving and I am getting little choppy and farting distorted noise. Something adds to sound :(. Is i t posiible to use Zaptel TE mode in 2.4 kernel? Thanks in advance for any help. Regards, Corvin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Bristuff and incoming call problems
I have exactly the same problem. It was also the same with RC3. It seems that after a couple of days of working fine, at some point incoming calls fail but outgoing calls still work (or I would hear user complaints earlier). For the lack of ring problem, I do the following in extensions.conf: [fromexternal] exten = s,1,Ringing exten = s,2,Wait,3 exten = s,3,Answer exten = s,4,Wait,1 exten = s,5,Background(enter-ext-of-person) So Asterisk singals a ringing tone for 3 seconds so that the caller's phone has a chance to ring, then answers and plays the 'enter the extension of the person you want to call' thing while at the same time listening for digits. I don't know if this is the right or expected approach but it works for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Friday, February 04, 2005 12:16 PM To: Asterisk Users List Subject: [Asterisk-Users] Bristuff and incoming call problems Hi list! I have some strange problems with Asterisk 1.0.5-BRIstuffed-0.2.0-RC5. Very regularly asterisk seems to lose connectivity with the ISDN line. If you try to call in you get the information tone that the number is not in use. Outbound calls do stil work however. Unloading the modules and reloading them and start/stop asterisk will solve the problem. Another problem that occurs regularly : When you make an inbound call to asterisk the calling party does not get the tone that the phone is ringing on the receiving end. The line just seems completely dead untill the phone is picked up and you can hear the other party. Is this an asterisk / bristuff problem or something for the telco to sort out? Who should generate the ringing signal to the calling party? Thanks!!! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Odd behaviour between Grandstream and Xlite
I allow them to us any codec except speex (which seems to crash Asterisk when used from an Xlite). But it would be good if the user could choose their preferred codec because with a softphone on a laptop sometimes you are on a connection with good bandwidth to Asterisk and sometimes somewhere with terrible bandwidth so you want to use a low bandwidth codec. If Asterisk chooses for you then the codec choosing feature on the Xlite is pointless. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, February 03, 2005 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Odd behaviour between Grandstream and Xlite Whatever codec I choose in Xlite, when calling the Grandstream it always uses the GSM codec even if it is greyed out. Whatever codec I choose in Xlite, when getting called by the Grandstream it always uses ulaw even if it is greyed out. and what about the phone config in sip.conf ? what codec do you allow them to use ? I think * doesn't care what codec is grayed out in X-lite, her use what sip.conf tell him he can hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Bristuff and incoming call problems
Sure you can put whatever you like in the answering config. That is just mine because after dialing an incoming number the caller dials an extension number that Asterisk uses to complete the call. Junhanns does post on this group occasionally so I guess he watches it but I haven't so far seen any useful messages on how to solve this problem. A behaviour I did notice once was that if you dialed in and then waited about 4 seconds then you did eventually get a ringing tone, which suggests that it was connecting but waiting for some timeout before following the context code; i.e. it was working but had a timeout or was working extremely slowly. Next time to thing behaves badly I will check if this is still the behaviour. If it doesn't get fixed then I will probably use a script that stops asterisk, reloads the modules, and starts asterisk again and runs it at say 5am every morning. Not ideal but what can you do? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Friday, February 04, 2005 2:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Bristuff and incoming call problems Thanks for the replies to my cry for help! :) The weird thing is that sometimes tyhe caller does hear the phone ringing, and sometimes the line is dead. I will try your workaround, will it also work without playing the message for an extension? I use it at home and it sounds a bit silly :) Are these bugs known at Junghanns? On Fri, 4 Feb 2005, Rob Scott wrote: I have exactly the same problem. It was also the same with RC3. It seems that after a couple of days of working fine, at some point incoming calls fail but outgoing calls still work (or I would hear user complaints earlier). For the lack of ring problem, I do the following in extensions.conf: [fromexternal] exten = s,1,Ringing exten = s,2,Wait,3 exten = s,3,Answer exten = s,4,Wait,1 exten = s,5,Background(enter-ext-of-person) So Asterisk singals a ringing tone for 3 seconds so that the caller's phone has a chance to ring, then answers and plays the 'enter the extension of the person you want to call' thing while at the same time listening for digits. I don't know if this is the right or expected approach but it works for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Friday, February 04, 2005 12:16 PM To: Asterisk Users List Subject: [Asterisk-Users] Bristuff and incoming call problems Hi list! I have some strange problems with Asterisk 1.0.5-BRIstuffed-0.2.0-RC5. Very regularly asterisk seems to lose connectivity with the ISDN line. If you try to call in you get the information tone that the number is not in use. Outbound calls do stil work however. Unloading the modules and reloading them and start/stop asterisk will solve the problem. Another problem that occurs regularly : When you make an inbound call to asterisk the calling party does not get the tone that the phone is ringing on the receiving end. The line just seems completely dead untill the phone is picked up and you can hear the other party. Is this an asterisk / bristuff problem or something for the telco to sort out? Who should generate the ringing signal to the calling party? Thanks!!! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 1.0.5-BRIstuffed-0.2.0-RC5 caller id?
I use
pritrustusercid = no
In zapata.conf and then it seems to work.
No idea if it is a bug or not or if this is a proper solution.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Tuesday, February 01, 2005 10:11 PM
To: Asterisk Users List
Subject: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 caller id?
I tried to get callerid working the normal way but the cid is never
passed to the phone.
It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in
extensions.conf
which I found in the wiki:
http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc
Is this intended behaviour, or still a bug?
It does work but it only shows one zero even though I have
nationalprefix = 0 internationalprefix = 00 in zapata.conf
I guess it should show a double zero because there is already a zero
prefix in the SetCIDNum(0${PRI_NETWORK_CID})?
I haven't received any international calls yet but will they not show up
with only one zero now?
Cheers!
Remco
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RE: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 caller id?
Also just adding
callerid=asreceived
To zapata.conf also seems to work.
Works for local or national calls where I am.
I don't know about international calls.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Tuesday, February 01, 2005 10:11 PM
To: Asterisk Users List
Subject: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 caller id?
I tried to get callerid working the normal way but the cid is never
passed to the phone.
It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in
extensions.conf
which I found in the wiki:
http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc
Is this intended behaviour, or still a bug?
It does work but it only shows one zero even though I have
nationalprefix = 0 internationalprefix = 00 in zapata.conf
I guess it should show a double zero because there is already a zero
prefix in the SetCIDNum(0${PRI_NETWORK_CID})?
I haven't received any international calls yet but will they not show up
with only one zero now?
Cheers!
Remco
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[Asterisk-Users] Odd behaviour between Grandstream and Xlite
Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as does the Grandstream Why the asymmetrical behaviour? Why does the Xlite accept a non-GSM call when it is set to do GSM? Thanks for any help. Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Odd behaviour between Grandstream and Xlite
Actually it is worse than that. Whatever codec I choose in Xlite, when calling the Grandstream it always uses the GSM codec even if it is greyed out. Whatever codec I choose in Xlite, when getting called by the Grandstream it always uses ulaw even if it is greyed out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Scott Sent: Thursday, February 03, 2005 7:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Odd behaviour between Grandstream and Xlite Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as does the Grandstream Why the asymmetrical behaviour? Why does the Xlite accept a non-GSM call when it is set to do GSM? Thanks for any help. Rob ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Soft phone sound quality help
I've tried setting the QoS settings on the card and using the Microsoft QoS packet scheduler, in all combinations, but no changes. I don't think these applications use QoS anyway. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, January 28, 2005 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Soft phone sound quality help I have a client that experienced quality problems and he said the resolution turned out to be the QoS option for the nic card (even though their backbone didn't support QoS). Try the softphones with and without QoS to hear the difference. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: New Firefly version
Also sound quality seems to be poor using the ULAW codec. I am using: - latest Firefly on Windows XP SP2 - Asterisk 1.0.5 patched coupled with Bristuff-0.2.0-RC5 with Florz patch for zaphfc - Linux kernel 2.6.9-1.681_FC3 Fedora Core 3 (obviously) - connecting to FWD dialing 411 info service Any other codec is better and useable. Clearly it seems to be optimized for iLBC. ULAW is unusable for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hhandresen Sent: Thursday, January 27, 2005 11:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: New Firefly version Hi Adam, Sory to say it, bu it still interupt the mouse if you have microsoft wireless mouse/keayboard. The mouse jumps around on the screen. Any news on this ? /HHA Adam Hart wrote: As always, I'm happy to announce a new version of Firefly. Firefly 1.9.8 has more of what you want and less of what you don't http://www.virbiage.com/firefly/download/firefly-thirdparty.exe There's a few bug fixes - notably fixed the Reject button and sending of audio before answering in some circumstances. -Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Soft phone sound quality help
Anyone got any tips on improving sound quality on soft phones running under Window XP SP2? I have tried Xlite, SJPhone and Firefly. They all seem to have significant sound quality problems. We have a reasonable sized network of several hundred devices connected together using Layer 2 switches, i.e. pretty dumb switches with no QoS. I also have a Grandstream connected to the same switching gear. The Grandstream sounds pretty good with very few drop outs or sound problems on ulaw. The soft phones all have problems although they get less when going to a lower bandwidth codec, but then lower bandwidth gives you worse sound quality too. Is there any way I can improve sound quality on the softphones? Or it is pretty well the general rule that they have poor sound quality? It makes sense to install a softphone on each of the 200 desktops we have but not to buy 200 Grandstreams or equivalent, and not to upgrade all our network switches. On the Asterisk side, jitter buffer is turned on with default settings. TOS is turned on for SIP although I doubt the switches can do anything with it. I have played around with a lot of Asterisk settings but without getting good results. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] zaphfc no callerid incoming to SIP phone butvisible in logfile
Try commenting out the line
pritrustusercid = yes
Or set it to 'no'.
That worked for me.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jens
Sent: Friday, January 21, 2005 7:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone
butvisible in logfile
Hi,
I think you didn't set usecallerid=yes in your zapata.conf?
Another way is to set the callerid in your extensions.conf via exten =
807440,2,SetCIDNum(0${CALLERIDNUM}). So you also have a 0 in front of the
displayed number - nice for callback.
regards
Jens
Hello,
I've added a ZAPHFC card to my CAPI based system. Calls coming in via
ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
via CAPI do forward the caller id to the SIP phones.
--
Jens Lentföhr
http://www.jens-it.de
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RE: [Asterisk-Users] chan_misdn 0.0.3-rc5 - new release ! Please testit.
What are the advantages in using mISDN over other solutions? If I knew why it was a good idea (like does it have better sound quality than alternatives?) then I would put the time in to test it, and also improve the Wiki. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Häger Sent: Friday, January 21, 2005 5:49 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] chan_misdn 0.0.3-rc5 - new release ! Please testit. Hi there, we've announced a new release of our chan_misdn channel driver. chan_misdn is a GPL channel driver for the new Linux ISDN-Layer mISDN (www.isdn4linux.org). So you can use all from mISDN supported ISDN catds in Asterisk. Feel free to donwload and test it at : http://www.beronet.com/download/chan_misdn-beta-0.0.3-rc5.tgz You can report bugs and feature requests to www.beronet.com/bugs Have fun! Thomas. -- *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Friedrichstr. 231 Haus D, 4. OG 10969 Berlin FON:+49 (0) 30 259389-14 FAX:+49 (0) 30 259389-19 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Poor sound quality on ISDN BRI calls
I've been struggling with connection Asterisk to ISDN BRI lines for a while. I have it working with the latest bristuff and compatible Asterisk version: Asterisk 1.0.3-BRIstuffed-0.2.0-RC3a I am using a cheap Centronics ISDN card and the zaphfc drivers. It works but users complain that the sound quality is not good. They have Xlite phones on their desktops. Xlite to Xlite through Asterisk is fine. Xlite to PSTN through ISDN is not good. Anyone got any experience with this kind of setup and improving sound quality? I will add anything new info to the Wiki. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound quality poor everywhichway
I am hoping someone is going to bite on the sound quality issue. I have Asterisk connected via a Conceptronics HFC-S card to an Elmeg ICT880 PBX internal extension line. Running Asterisk 1.0.3 and latest Bristuff. I have firefly and Xlite clients running on Windows XP. Calls between Xlites through Asterisk seem to be fine. Calls from anything to ISDN is terrible whatever settings I change. Calls on VoiP for example to the 411 service on FWD through my Asterisk box are not as good as using the same service by phone, not by a long way. There is distortion and also a kind of harmonic tone in the background when the person is talking on the other end (i.e. it sound like someone is playing a low toned kazoo in time to the person talking). Everything is going ulaw. It is on a quiet network. The Desktop is an IBM ThinkCenter with built in audio - 2.66 GHz P4. Anyone got any ideas? I am pulling my hair out here trying to get this working. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is an unregistered phone busy?
Asterisk seems to regard an unregistered phone to be busy. Is that correct? Is not an unregistered phone unavailable? It is odd to me that if someone dials an unregistered extension, then the dialplan jumps to busy and voicemail kicks in saying that the person is on the phone, when clearly they can't be if the phone hasn't registered. Any way around this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sounds cut out problem - HFC-S card, zaphfc, Xlite
Hello Asteriskians! I have an Asterisk box with a simple HFC card in it and a bunch of people using the Xlite software to connect. The HFC card is connected to an internal extension on our legacy PBX. So far so good. The Xlite clients can call each other, and the internal extensions on the PBX and the Xlites can call each other, no problem. The problem is when using an Xlite to dial an external number through the legacy PBX. What seems to happen is that there is some kind of noise suppression so that unless the remote party is speaking very loudly the sound cuts out. Now, I don't know if it is the ISDN connection, Asterisk or the Xlite client that is causing the problem. I've tried different settings on everything I can think of and trawled the web for days but so far nothing useful. I've turned off silence suppression on all the Xlites. I've turned up the rxgain on the ISDN channel in case it is too quiet. Nothing so far has helped. Calling directly through the PBX from a normal extension phone doesn't seem to have any problems. Anyone have any idea what I should look at? Thanks Rob Scott EPAM Systems Ltd. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Devkit Lite setup
At one point I did have this kit working but since upgrading to the latest Asterisk, it no longer seems to. I had the following problems after several reinstallations: - USB adaptor had a proper dialtone, asterisk recognised the pickup, but pressing keys on the handset had no effect - USB adaptor produced a strange horrible tone, not a dialtone; pressing keys has no effect - USB adaptor silent; pickup not recognised by Asterisk So you can see my situtation has gradually deteriorated with each tinkering/CVS-reinstall of the system. What I don't understand is what Linux or other elements do I need to get the USB adaptor to work? I am running RedHat 9 with the 2.4.20-20.9 kernel. Apart from these problems, I also noticed the following when using the USB adaptor: - unplugging and plugging the handset into the line side of the adaptor killed the adaptor; have to restart the machine to clear the problem. - unplugging the USB adaptor from the USB port causes all manner of problems. 'rmmod'-ing and 'insmod'-ing astrisk related modules doesn't help. Running asterisk or 'ztcfg' always complains of problems on the USB adaptor channel. Restart of machine needed. - sometimes problems with the computer's audio system ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: 21 August 2003 21:01 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE:911, networks of * servers, etc. (was: VOIP Dialtone?) Yes, I'm familiar with the E911 platforms and their requirements to some degree. The trick is that the people running Asterisk PBX systems have no visibility into SS7, and that is an unreasonable expectation, so some other out-of-band method for moving caller location to the PSAP is required. As far as geographic location tracking is concerned: that is the user's problem. If they don't have the correct information in their device, then they're SOL. There is _no way_ to develop lat/lon/alt coordinates from an IP address, despite what any .com flash-in-the-pan company says they can do with their clever databases. Thus, the PBX/switch provider will have to enforce their own database of device-to-geographic-coordinates. (As mentioned, maybe a SIP header is a reasonable thing to use for the UA to relay this data to the proxy.) I am not concerned so much about the ability of the devices to send their data to the proxy: I am VERY concerned about how the proxy then looks up the appropriate PSAP, and then relays the data for the call to that PSAP. JT 911 through the phone system is tricky business. e911 which is the automated process of handing the address to the 911 center uses the SS7 database to do it's work (the database is created when the LEC runs physical lines to locations not by people filling anything out). Cell phone service providers have the simuliar problems as VoIP service providers are facing are realizing with call forwarding and call following it will get worse.. Congress has mandated that the cell phone industry make it possible to track a cell phone users within 300yards via cell sites and triangulation. By 2005 every cell phone will be required to have a GPS and send GPS information to the 911 system when they call 911. If you want more information on e911 try http://www.fcc.gov/911/enhanced/ . As the cell phone industry grows there will be a need for a national 911 call routing center. I bet it won't be free. Original Message: - From: John Todd [EMAIL PROTECTED] Date: Thu, 21 Aug 2003 01:32:24 -0700 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 911, networks of * servers, etc. (was: VOIP Dialtone?) OK, that VOIP dialtone? thread was getting really out of hand, so I'll condense my answers into one big ugly message: 1) 911 service. Yes, that is one of three reasons to keep your PSTN line. The other two reasons are: Inbound calls from local callers still should work on a POTS line, for now. You can't find VOIP providers in most area codes, so you'll most likely need to have a local number that finds it's way to you for local tasks. Secondly, the Internet is not as reliable as the phone system. Sorry, folks, it just works that way right now despite what your network engineer might tell you. That's not to say it's unreliable, but those last two nines are very expensive... Besides, any good network engineer will tell you that you should have multiple paths for your IP connectivity. With few exceptions, most homes do not have multipath connectivity. (note: businesses may in fact have better uptime on their IP network than their phone network, if they have competent engineers and a reasonable budget.) 1.5) There are reasonable technical solutions to this problem, but for the life of me I can't figure out why the 911 centers haven't gotten their act together and solved this. There are two halves to this problem: What PSAP do I call? (and what phone number) and How do I get my location data to the PSAP once I call them? C'mon, this is not difficult. The first question can be answered trivially: there _must_ be a database of address-to-PSAP mappings. Any PBX administrator (or SIP phone owner, for that matter) should be able to figure out their address. Methods for associating the PSAP number with the phone are numerous, and trivially implemented - if people don't keep their address information updated, they're SOL (though you can remind them in an automated fashion to keep it updated - just forbid them from using the service unless they verify the address every month or so.) The second question is more difficult, but certainly possible. There may be kludge ways of doing it, and there should be more elegant ways of doing it. A SIP header with lat/lon/alt data that gets sent from the UA only on 911 (or other programmable string) calls might be reasonably elegant... maybe. But that only gets the data to the SIP proxy. That doesn't solve the issue of how you get that data from the SIP proxy to the PSAP, which at some point will be almost certainly through a PSTN connection... ADSI FSK, maybe? Ugly, and PSAPs would not want to invest in equipment. A national
