RE: [Asterisk-Users] Question about remote POTS lines
You might look at installing FXS adapters at the remote sites which would take a call from the C.O. and then pass it to the Asterisk system at the Main site. Then you could either use the SIP phones or IAXy adapters at the remote sites. This would in essence terminate all the lines for all offices in the Main office and then you can write the dial plan accordingly. Tim. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Dossey Sent: Monday, November 15, 2004 3:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Question about remote POTS lines I have a client who asked me about a situation they have. They have a main office and 3 remote offices. We are installing an Asterisk server at the main office with SIP phones in the remotes. Each remote office only has 1 person. The remote offices currently have a POTS line that has a published number. They want to keep that number. The problem is that they would like to somehow link those remote POTS lines back to the main office, so people in the main office can answer their calls when they are away. They could install an asterisk server in those remote offices and link them back to the main office, but that seems like overkill for a single POTS line. Any ideas? ___ 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] Question about remote POTS lines
I have a client who asked me about a situation they have. They have a main office and 3 remote offices. We are installing an Asterisk server at the main office with SIP phones in the remotes. Each remote office only has 1 person. The remote offices currently have a POTS line that has a published number. They want to keep that number. The problem is that they would like to somehow link those remote POTS lines back to the main office, so people in the main office can answer their calls when they are away. They could install an asterisk server in those remote offices and link them back to the main office, but that seems like overkill for a single POTS line. Any ideas? ___ 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] Question about remote POTS lines
How remote are the remote offices? Miles? States? Countries? Best of my knowledge, the days of exchanges based on proximity to a particular CO are over, and those numbers (assuming they are in the same area code) often can be routed anywhere. You could also look into having a company like VoicePulse take over the PSTN termination and shoot you a VoIP link to the central * server. Greg Jim Dossey wrote: I have a client who asked me about a situation they have. They have a main office and 3 remote offices. We are installing an Asterisk server at the main office with SIP phones in the remotes. Each remote office only has 1 person. The remote offices currently have a POTS line that has a published number. They want to keep that number. The problem is that they would like to somehow link those remote POTS lines back to the main office, so people in the main office can answer their calls when they are away. They could install an asterisk server in those remote offices and link them back to the main office, but that seems like overkill for a single POTS line. Any ideas? ___ 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] Question about remote POTS lines
How remote are the remote offices? Miles? States? Countries? Best of my knowledge, the days of exchanges based on proximity to a particular CO are over, and those numbers (assuming they are in the same area code) often can be routed anywhere. You could also look into having a company like VoicePulse take over the PSTN termination and shoot you a VoIP link to the central * server. I have a client who asked me about a situation they have. They have a main office and 3 remote offices. We are installing an Asterisk server at the main office with SIP phones in the remotes. Each remote office only has 1 person. The remote offices currently have a POTS line that has a published number. They want to keep that number. The problem is that they would like to somehow link those remote POTS lines back to the main office, so people in the main office can answer their calls when they are away. They could install an asterisk server in those remote offices and link them back to the main office, but that seems like overkill for a single POTS line. hmm maybe they should have the sipura 3000 out there analog phones You can config it so that after so many rings on the pstn fxo port it fall over to the call fwd sip addr, or you can train the remote off to do *72 to fwd all calls Or you can set up a small web page to post the same info as the sipura config page that allows call fwding also a sipura gives the local office redundancy in case the inet line goes down ___ 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] Question about remote POTS lines
Hoi about having the calls forwarded by your phone company? Usually you can dial *21*number# or something and your calls go to a remote party. Same goes for delayed forwarding *61* Rene Kluwen Chimit - Original Message - From: Jim Dossey To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, November 15, 2004 11:57 PM Subject: [Asterisk-Users] Question about remote POTS lines I have a client who asked me about a situation they have. They have a main office and 3 remote offices. We are installing an Asterisk server at the main office with SIP phones in the remotes. Each remote office only has 1 person. The remote offices currently have a POTS line that has a published number. They want to keep that number. The problem is that they would like to somehow link those remote POTS lines back to the main office, so people in the main office can answer their calls when they are away. They could install an asterisk server in those remote offices and link them back to the main office, but that seems like overkill for a single POTS line.Any ideas? ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] question about asterisk
People, some people like pixelFriend and Jonathan Augenstine, send me references about asterisk documentation, I read this and ok thanks :), my project is: I have two office of construction: office_1 and office_2 I want connect the two offices by dedicated line, and connect the two analog PBX of this offices and transfer VoIP between two offices. Ok one time that of two offices connected, I want install of telephone IP on both office and CALL. And I want call from office_1 to office_2 and office_2 switch this call to PBX for talk with mi house. For this situations what CARDS I need? but I see too many CARDS from digium :) and I a newbie on this. Thanks for the help or link for read... :) , Olger Merlos Valverde Programas Integrados de America S.A Mensaje enviado y revisado de Virus por nuestro servidor de Correo. Ekstrom S.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] question about asterisk
On Thu, 28 Oct 2004 20:05:03 -0600, Olger Merlos Valverde [EMAIL PROTECTED] wrote: I have two office of construction: office_1 and office_2 I want connect the two offices by dedicated line, and connect the two analog PBX of this offices and transfer VoIP between two offices. Ok one time that of two offices connected, I want install of telephone IP on both office and CALL. And I want call from office_1 to office_2 and office_2 switch this call to PBX for talk with mi house. For this situations what CARDS I need? but I see too many CARDS from digium :) and I a newbie on this. at minimum you will need one card: Wildcard X100P ... [IPphone-Office1] | SIP/LAN | [Aterisk-Office1] | IAX/WAN | [Asterisk-Office2]---[Wildcard-X100P]---analog---[PBX-Office2]--PSTN | SIP/LAN | [IPphone-Office2] SIP IP phones are configured in /etc/asterisk/sip.conf Wildcard X100P card is configured in /etc/zaptel.conf and /etc/asterisk/zapata.conf IAX link between two Asterisk servers is configured in /etc/asterisk/iax.conf Also relevant: /etc/asterisk/extensions.conf for dialplan use all of the above (sip.conf, iax.conf, etc etc) as keywords for search at http://www.voip-info.org for details. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Question about ISDN reason codes
All, newbie to Asterisk and just trying to get a load of bits together including PSTN interface using Digium Quad E1 interfaces using EuroISDN. Question can I/how do I get access to the ISDN reason codes for call disconnect? The purpose is to be able to make routing choices based on the reason code returned from a PRI connected interface/channel. For example if the reason code indicates say network congestion or no route then I want to be able to try an alternative route by manipulating the digit string and retrying on another interface, for example to a SIP ITSP. Here's the scenario - call originates on IP cloud from an IP Phone/IAD/ATA connected via a proxy server (SER), destination PSTN via Asterisk, First choice route is out of an ISDN PRI port on the Digium board, if this route is either congested - no channels left to dial out on or the far end network has a problem such as can't route to that destination, Then take a second, third etc choice of route to that destination say via a SIP connection to another ITSP, or another PRI Interface connected to another PSTN provider. Any suggestions, thoughts, configuration examples and general advice welcome! Neill;o) ___ 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] Question about ISDN reason codes
Neill Wilkinson wrote: All, newbie to Asterisk and just trying to get a load of bits together including PSTN interface using Digium Quad E1 interfaces using EuroISDN. Question can I/how do I get access to the ISDN reason codes for call disconnect? /path/to/asterisk/docs/README.variables Pay special attention to DIALSTATUS and to HANGUPCAUSE. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ 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] Question about ISDN reason codes
Did someone actually got this to work ? The examples on voip-info are not correct i think. It will never go to priority 2, since the call already got hungup at that point, using dial|g doesnt seem to work either. Joachim At 04:17 22/10/2004, you wrote: Neill Wilkinson wrote: All, newbie to Asterisk and just trying to get a load of bits together including PSTN interface using Digium Quad E1 interfaces using EuroISDN. Question can I/how do I get access to the ISDN reason codes for call disconnect? /path/to/asterisk/docs/README.variables Pay special attention to DIALSTATUS and to HANGUPCAUSE. ___ 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] Question about ISDN reason codes
Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay special attention to the [macro-dial-result] joachim wrote: Did someone actually got this to work ? The examples on voip-info are not correct i think. It will never go to priority 2, since the call already got hungup at that point, using dial|g doesnt seem to work either. Joachim At 04:17 22/10/2004, you wrote: Neill Wilkinson wrote: All, newbie to Asterisk and just trying to get a load of bits together including PSTN interface using Digium Quad E1 interfaces using EuroISDN. Question can I/how do I get access to the ISDN reason codes for call disconnect? /path/to/asterisk/docs/README.variables Pay special attention to DIALSTATUS and to HANGUPCAUSE. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ 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] Question about ISDN reason codes
Thnx Eric, Could you also post your extensions.conf ? (or just the extension calling that macro ?) Joachim At 04:48 22/10/2004, you wrote: Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay special attention to the [macro-dial-result] joachim wrote: Did someone actually got this to work ? The examples on voip-info are not correct i think. It will never go to priority 2, since the call already got hungup at that point, using dial|g doesnt seem to work either. Joachim At 04:17 22/10/2004, you wrote: Neill Wilkinson wrote: All, newbie to Asterisk and just trying to get a load of bits together including PSTN interface using Digium Quad E1 interfaces using EuroISDN. Question can I/how do I get access to the ISDN reason codes for call disconnect? /path/to/asterisk/docs/README.variables Pay special attention to DIALSTATUS and to HANGUPCAUSE. ___ 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] Question about ISDN reason codes
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
exten = _91NXXNXX,2,Macro(dial-result)
joachim wrote:
Thnx Eric,
Could you also post your extensions.conf ? (or just the extension
calling that macro ?)
Joachim
At 04:48 22/10/2004, you wrote:
Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay
special attention to the [macro-dial-result]
joachim wrote:
Did someone actually got this to work ?
The examples on voip-info are not correct i think.
It will never go to priority 2, since the call already got hungup at
that point, using dial|g doesnt seem to work either.
Joachim
At 04:17 22/10/2004, you wrote:
Neill Wilkinson wrote:
All,
newbie to Asterisk and just trying to get a load of bits together
including PSTN interface using Digium Quad E1 interfaces using
EuroISDN.
Question can I/how do I get access to the ISDN reason codes for
call disconnect?
/path/to/asterisk/docs/README.variables
Pay special attention to DIALSTATUS and to HANGUPCAUSE.
begin:vcard
fn:Eric Wileing
n:Wileing;Eric
email;internet:[EMAIL PROTECTED]
tel;work:504-899-1387 x2120
x-mozilla-html:FALSE
version:2.1
end:vcard
___
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] Question about ISDN reason codes
Are you sure this works ? (and does it work whatever end hung up ?)
If it works, its not expected behaviour. (at least i dont think it is, it
should never go to the next priority when the call got hungup).
zoa.
At 05:06 22/10/2004, you wrote:
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
exten = _91NXXNXX,2,Macro(dial-result)
joachim wrote:
Thnx Eric,
Could you also post your extensions.conf ? (or just the extension calling
that macro ?)
Joachim
At 04:48 22/10/2004, you wrote:
Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay
special attention to the [macro-dial-result]
joachim wrote:
Did someone actually got this to work ?
The examples on voip-info are not correct i think.
It will never go to priority 2, since the call already got hungup at
that point, using dial|g doesnt seem to work either.
Joachim
At 04:17 22/10/2004, you wrote:
Neill Wilkinson wrote:
All,
newbie to Asterisk and just trying to get a load of bits together
including PSTN interface using Digium Quad E1 interfaces using EuroISDN.
Question can I/how do I get access to the ISDN reason codes for call
disconnect?
/path/to/asterisk/docs/README.variables
Pay special attention to DIALSTATUS and to HANGUPCAUSE.
___
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] Question about ISDN reason codes
-Original Message-
From: joachim [mailto:[EMAIL PROTECTED]
Sent: Friday, October 22, 2004 6:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Question about ISDN reason codes
Are you sure this works ? (and does it work whatever end hung up ?)
If it works, its not expected behaviour. (at least i dont
think it is, it
should never go to the next priority when the call got hungup).
zoa.
At 05:06 22/10/2004, you wrote:
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
exten = _91NXXNXX,2,Macro(dial-result)
Check out the current config/extensions.conf.sample. This is exactly
How the relatively new dialstatus variable is used.
Robert Jackson
(Excerpt from extensions.conf.sample):
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20
seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to
start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail
w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #, return to
start
exten = _s-.,1,Goto(s-NOANSWER,1); Treat anything else as no
answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the
user into VoicemailMain
___
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] Question about ISDN reason codes
Yes it works. It will go to priority 2 if the call was NOT ANSWERED for
any reason (busy, number not in service, etc). You may need to add ,,g
on the Dial line to get Asterisk to go to priority two if the CALLEE
hangs up.
I do not do post call processing if the CALLER hangs up.
joachim wrote:
Are you sure this works ? (and does it work whatever end hung up ?)
If it works, its not expected behaviour. (at least i dont think it is,
it should never go to the next priority when the call got hungup).
zoa.
At 05:06 22/10/2004, you wrote:
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
exten = _91NXXNXX,2,Macro(dial-result)
joachim wrote:
Thnx Eric,
Could you also post your extensions.conf ? (or just the extension
calling that macro ?)
Joachim
At 04:48 22/10/2004, you wrote:
Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay
special attention to the [macro-dial-result]
joachim wrote:
Did someone actually got this to work ?
The examples on voip-info are not correct i think.
It will never go to priority 2, since the call already got hungup
at that point, using dial|g doesnt seem to work either.
Joachim
At 04:17 22/10/2004, you wrote:
Neill Wilkinson wrote:
All,
newbie to Asterisk and just trying to get a load of bits together
including PSTN interface using Digium Quad E1 interfaces using
EuroISDN.
Question can I/how do I get access to the ISDN reason codes for
call disconnect?
/path/to/asterisk/docs/README.variables
Pay special attention to DIALSTATUS and to HANGUPCAUSE.
begin:vcard
fn:Eric Wileing
n:Wileing;Eric
email;internet:[EMAIL PROTECTED]
tel;work:504-899-1387 x2120
x-mozilla-html:FALSE
version:2.1
end:vcard
___
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] Question about ISDN reason codes
Aha, oke :)
I was thinking of the answered statuses. That g was not working for me last
time i checked.
But so at least its working when a call did not get answered, thats already
good news for me.
Thanks a lot...
Joachim
At 05:23 22/10/2004, you wrote:
Yes it works. It will go to priority 2 if the call was NOT ANSWERED for
any reason (busy, number not in service, etc). You may need to add ,,g on
the Dial line to get Asterisk to go to priority two if the CALLEE hangs up.
I do not do post call processing if the CALLER hangs up.
joachim wrote:
Are you sure this works ? (and does it work whatever end hung up ?)
If it works, its not expected behaviour. (at least i dont think it is, it
should never go to the next priority when the call got hungup).
zoa.
At 05:06 22/10/2004, you wrote:
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
exten = _91NXXNXX,2,Macro(dial-result)
joachim wrote:
Thnx Eric,
Could you also post your extensions.conf ? (or just the extension
calling that macro ?)
Joachim
At 04:48 22/10/2004, you wrote:
Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay
special attention to the [macro-dial-result]
joachim wrote:
Did someone actually got this to work ?
The examples on voip-info are not correct i think.
It will never go to priority 2, since the call already got hungup at
that point, using dial|g doesnt seem to work either.
Joachim
At 04:17 22/10/2004, you wrote:
Neill Wilkinson wrote:
All,
newbie to Asterisk and just trying to get a load of bits together
including PSTN interface using Digium Quad E1 interfaces using EuroISDN.
Question can I/how do I get access to the ISDN reason codes for
call disconnect?
/path/to/asterisk/docs/README.variables
Pay special attention to DIALSTATUS and to HANGUPCAUSE.
___
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] Question about ISDN reason codes
On Fri, 22 Oct 2004, joachim wrote: I was thinking of the answered statuses. That g was not working for me last time i checked. Can you post your Dial line (and preferably the lines after that as well)? The 'g' option should work. It does for us, but we are a bit behind HEAD. Peter ___ 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] question about type=user in sip.conf
Hi, I may be missing something here, but I don't really understand how asterisk supposed to handle type=user. Suppose I have the following config (mostly taken from default sip.conf.sample): sip.conf: context=sip ;default context for incoming calls ... register = [EMAIL PROTECTED] .. [sip-proxy-out] type=peer username=user secret=secret .. [sip-proxy] type=user context=from-proxy The question is how asterisk determines that the call is from sip-proxy? Whatever I do all incoming calls coming from sip-proxy (or from any other sip device not registered locally) get into sip context (default context) instead of sip-proxy context. Could anybody enlighten me on this or point out to some documentation? Thanks, Michael ___ 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] Question/Future Request for Call Queues
I have some quetions/ideas for the Asterisk Call Queues system. System information: - Fedora Core 1 - Kernel 2.4.22-1.2115.nptl - Asterisk CVS-HEAD-09/08/04-17:43:15 1. I sould like it that if a user is in the que and the expected wait time is longer then xxx seconds or there are more then xxx callers. That there is played a sound from directory xxx with some product information (advertisement) 2. You can specify a member sequence with an agument on the memeber function like this: member = SIP/user,1 ;(ringing with first attempt) member = SIP/someuser,2 ;(ringing with first attempt) member = SIP/otheruser,3 ;(ringing with first attempt) member = SIP/someotheruser,3 ;(ringing with first attempt) But this sequence is not working as I aspected. I aspected that it's working like: member = SIP/user,1 ;(ringing with first attempt) member = SIP/someuser,2 ;(ringing with second attempt) member = SIP/otheruser,3 ;(ringing with third attempt) member = SIP/someotheruser,3 ;(ringing with third attempt) My strategy is currently set on ringall, beceuse the other options does not specify the option I like to have. Regards, 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] Question about the 'fax' extension
I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk Paul Seniuk.vcf Description: Binary data ___ 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] Question about the 'fax' extension
If you use extension dedicated to fax, then you don't need to use the fax extenstion, but just call the rxfax application directly as you would call the answer application exten = 123456,1,rxfax(...) But of course you may just use different fax extensions for different contexts. Regards, Marc [EMAIL PROTECTED] wrote: I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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] Question about the 'fax' extension
What happens if I want it to work over the same DiD though? Does Answer() take care of this? How do I jump to the fax extension if it detects a faxtone? Paul Seniuk -Original Message- From: mstorck [mailto:[EMAIL PROTECTED] Sent: September 20, 2004 3:51 PM To: asterisk-users Subject: Re: [Asterisk-Users] Question about the 'fax' extension If you use extension dedicated to fax, then you don't need to use the fax extenstion, but just call the rxfax application directly as you would call the answer application exten = 123456,1,rxfax(...) But of course you may just use different fax extensions for different contexts. Regards, Marc [EMAIL PROTECTED] wrote: I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk -- -- ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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] Question about the 'fax' extension
Answer() will jump to the fax extension in the same context just automatically... Marc [EMAIL PROTECTED] wrote: What happens if I want it to work over the same DiD though? Does Answer() take care of this? How do I jump to the fax extension if it detects a faxtone? Paul Seniuk -Original Message- From: mstorck [mailto:[EMAIL PROTECTED] Sent: September 20, 2004 3:51 PM To: asterisk-users Subject: Re: [Asterisk-Users] Question about the 'fax' extension If you use extension dedicated to fax, then you don't need to use the fax extenstion, but just call the rxfax application directly as you would call the answer application exten = 123456,1,rxfax(...) But of course you may just use different fax extensions for different contexts. Regards, Marc [EMAIL PROTECTED] wrote: I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk -- -- ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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] Question about the 'fax' extension
Ok, hopefully this allows me to have mutliple context's that each have their own fax extension. I will see if this works. Cheers, Pauly -Original Message- From: mstorck [mailto:[EMAIL PROTECTED] Sent: Monday, September 20, 2004 3:59 PM To: asterisk-users Subject: Re: [Asterisk-Users] Question about the 'fax' extension Answer() will jump to the fax extension in the same context just automatically... Marc [EMAIL PROTECTED] wrote: What happens if I want it to work over the same DiD though? Does Answer() take care of this? How do I jump to the fax extension if it detects a faxtone? Paul Seniuk -Original Message- From: mstorck [mailto:[EMAIL PROTECTED] Sent: September 20, 2004 3:51 PM To: asterisk-users Subject: Re: [Asterisk-Users] Question about the 'fax' extension If you use extension dedicated to fax, then you don't need to use the fax extenstion, but just call the rxfax application directly as you would call the answer application exten = 123456,1,rxfax(...) But of course you may just use different fax extensions for different contexts. Regards, Marc [EMAIL PROTECTED] wrote: I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk -- -- ___ 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 -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ 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] Question calling number
Hello all, I have a question concerning the calling number with an incoming PSTN call through a E100P : Here is what I see with a pri debug : Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '333007' ] [6c 0b 20 83 32 34 37 33 33 33 30 33 30] Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: Unknown Number Plan (0) Presentation: Presentation allowed of network provided number (3) '33030' ] [70 05 81 34 32 34 33] There are two calling numbers : one is the real number (the first one), the second one is the group number (I don't know the correct word for this, maybe head number of the group ?). Asterisk is using the second one when I think it should use the first one. How could I tell asterisk to pick the first number ? The info is decoded by libpri, so it should be possible to change asterisk's default behaviour. Thanks a lot ! Guillaume ___ 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] Question on echo's for Canadian Asterisk users ...
Thanks for the tips ... Like you said, dealing with carrier is not going to get me anywhere. The only thing GT recommended was grounding the server chasis :P I turned the echo cancellation with the same parameters you used and It doesnt even make a difference. I dug further into the zaptel/Makefile To find the the echo cancellation algoriths: #KFLAGS+=-DECHO_CAN_STEVE KFLAGS+=-DECHO_CAN_STEVE2 #KFLAGS+=-DECHO_CAN_MARK None of these were listed at all in the Makefile, so I added them And tried a recompile. Still a bad echo. It is like the echo cancellation Is not even working. Is there a way to verify its active or not? Cheers, Paul Seniuk -Original Message- From: akohlsmith-asterisk [mailto:[EMAIL PROTECTED] Sent: September 4, 2004 3:51 PM To: asterisk-users Subject: Re: [Asterisk-Users] Question on echo's for Canadian Asterisk users ... On Saturday 04 September 2004 16:57, [EMAIL PROTECTED] wrote: Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). I have a PRI with Bell Canada in Listowel, ON (519-291-). I have echo on some calls but not all -- it doesn't seem to have anything to do with what switch it's terminating on. Calling anywhere in Fordwich echos rather badly as do some Toronto numbers. The echo occurs on incoming and outgoing calls. (we only call out on the PRI for local, 800 and fax numbers). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. All that PRI gives you is one less hybrid in the circuit. That's it. NOC at GT is telling us that there is no echo cancellation enabled on this PRI. 'Talk to your rep' was the response I got To me thats crap, because they shouldnt be selling PRI's without this essential feature. Depending on who you talk to you will hear responses like 1. I have no idea what you're talking about. 2. We don't have echo cancellation hardware available on any PRI. 3. You must specifically provision the PRI with echo cancellation. I've found acceptable echo cancellation on the PRI with Asterisk's echo cancellation software on the TE405P with the following: - agressive cancellation echocancel=yes echocancelwhenbridged=yes echotraining=500 No need to worry about the echo canceller killing fax/data connections since just like the real echo cancellation hardware, asterisk will disable the echo cancel routines when it hears the correct disable tone on the line. You'll see something like zaptel Disabled echo canceller because of tone (tx) on channel 13. We were really having a lot of echo troubles but 20040831 CVS HEAD seems to have really helped, although it was certainly acceptable with 20040806 CVS HEAD. I haven't been able to locate a good hardware echo canceller on ebay yet (I keep missing the auctions). :-) -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 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] Question on echo's for Canadian Asterisk users ...
On Wednesday 08 September 2004 13:39, [EMAIL PROTECTED] wrote: None of these were listed at all in the Makefile, so I added them And tried a recompile. Still a bad echo. It is like the echo cancellation Is not even working. Is there a way to verify its active or not? It's not in the makefile, it's in zconfig.h. I've attached mine (which seems to work just fine). Hoping this doesn't break too many list rules, there are others who might benefit from a zconfig.h that seems to work very well. -A. /* * Zaptel configuration options * */ #ifndef _ZCONFIG_H #define _ZCONFIG_H #ifdef __KERNEL__ #include linux/config.h #include linux/version.h #endif /* Zaptel compile time options */ /* * Uncomment to disable calibration and/or DC/DC converter tests * (not generally recommended) */ /* #define NO_CALIBRATION */ /* #define NO_DCDC */ /* * Boost ring voltage (Higher ring voltage, takes more power) */ /* #define BOOST_RINGER */ /* * Define CONFIG_CALC_XLAW if you have a small number of channels and/or * a small level 2 cache, to optimize for few channels * */ /* #define CONFIG_CALC_XLAW */ /* * Define if you want MMX optimizations in zaptel * * Note: CONFIG_ZAPTEL_MMX is generally incompatible with AMD * processors and can cause system instability! * */ #define CONFIG_ZAPTEL_MMX /* * Pick your echo canceller: MARK2, MARK3, STEVE, or STEVE2 :) */ /* #define ECHO_CAN_STEVE */ /* #define ECHO_CAN_STEVE2 */ /* #define ECHO_CAN_MARK */ #define ECHO_CAN_MARK2 /* #define ECHO_CAN_MARK3 */ /* * Uncomment for aggressive residual echo supression under * MARK2 echo canceller */ /* #define AGGRESSIVE_SUPPRESSOR */ /* * Define to turn off the echo canceler disable tone detector, * which will cause zaptel to ignore the 2100 Hz echo cancel disable * tone. */ /* #define NO_ECHOCAN_DISABLE */ /* udev support */ #if LINUX_VERSION_CODE = KERNEL_VERSION(2,6,0) #define CONFIG_ZAP_UDEV #endif /* We now use the linux kernel config to detect which options to use */ /* You can still override them below */ #if defined(CONFIG_HDLC) || defined(CONFIG_HDLC_MODULE) /* #define CONFIG_ZAPATA_NET */ /* NEVER implicitly turn on ZAPATA_NET */ #if LINUX_VERSION_CODE = KERNEL_VERSION(2,4,20) #define CONFIG_OLD_HDLC_API #else #if LINUX_VERSION_CODE = KERNEL_VERSION(2,4,23) /* Starting with 2.4.23 the kernel hdlc api changed again */ /* Now we have to use hdlc_type_trans(skb, dev) instead of htons(ETH_P_HDLC) */ #define ZAP_HDLC_TYPE_TRANS #endif #if LINUX_VERSION_CODE = KERNEL_VERSION(2,6,3) #define HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGHT #endif #endif #endif #ifdef CONFIG_PPP #define CONFIG_ZAPATA_PPP #endif /* * Uncomment CONFIG_ZAPATA_NET to enable SyncPPP, CiscoHDLC, and Frame Relay * support. */ /* #define CONFIG_ZAPATA_NET */ /* * Uncomment CONFIG_OLD_HDLC_API if your are compiling with ZAPATA_NET * defined and you are using the old kernel HDLC interface (or if you get * an error about ETH_P_HDLC while compiling). */ /* #define CONFIG_OLD_HDLC_API */ /* * Uncomment for Generic PPP support (i.e. ZapRAS) */ /* #define CONFIG_ZAPATA_PPP */ /* * Uncomment to enable watchdog to monitor if interfaces * stop taking interrupts or otherwise misbehave */ /* #define CONFIG_ZAPTEL_WATCHDOG */ /* Tone zone info */ #define DEFAULT_TONE_ZONE 0 /* * Uncomment for Non-standard FXS groundstart start state (A=Low, B=Low) * particularly for CAC channel bank groundstart FXO ports. */ /* #define CONFIG_CAC_GROUNDSTART */ #endif ___ 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] Question on echo's for Canadian Asterisk users ...
Yeah it looks to be the same setup as mine I am going to try out Mark3 and the Aggressive Suppresor as well. Paul Seniuk -Original Message- From: akohlsmith-asterisk [mailto:[EMAIL PROTECTED] Sent: September 8, 2004 12:34 PM To: asterisk-users Subject: Re: [Asterisk-Users] Question on echo's for Canadian Asterisk users ... On Wednesday 08 September 2004 13:39, [EMAIL PROTECTED] wrote: None of these were listed at all in the Makefile, so I added them And tried a recompile. Still a bad echo. It is like the echo cancellation Is not even working. Is there a way to verify its active or not? It's not in the makefile, it's in zconfig.h. I've attached mine (which seems to work just fine). Hoping this doesn't break too many list rules, there are others who might benefit from a zconfig.h that seems to work very well. -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] Question on echo's for Canadian Asterisk users ...
On Wednesday 08 September 2004 14:46, [EMAIL PROTECTED] wrote: Yeah it looks to be the same setup as mine I am going to try out Mark3 and the Aggressive Suppresor as well. I'm using Mark2 and *no* agressive supressor (which surprised me, I thought I had it in there) -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] Question on echo's for Canadian Asterisk users ...
Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us that there is no echo cancellation enabled on this PRI. 'Talk to your rep' was the response I got To me thats crap, because they shouldnt be selling PRI's without this essential feature. Has anyone had any similar 'Canadian' experiences with this? Cheers, Paul Seniuk Paul Seniuk.vcf Description: Binary data ___ 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] Question on echo's for Canadian Asterisk users ...
On Saturday 04 September 2004 16:57, [EMAIL PROTECTED] wrote: Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). I have a PRI with Bell Canada in Listowel, ON (519-291-). I have echo on some calls but not all -- it doesn't seem to have anything to do with what switch it's terminating on. Calling anywhere in Fordwich echos rather badly as do some Toronto numbers. The echo occurs on incoming and outgoing calls. (we only call out on the PRI for local, 800 and fax numbers). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. All that PRI gives you is one less hybrid in the circuit. That's it. NOC at GT is telling us that there is no echo cancellation enabled on this PRI. 'Talk to your rep' was the response I got To me thats crap, because they shouldnt be selling PRI's without this essential feature. Depending on who you talk to you will hear responses like 1. I have no idea what you're talking about. 2. We don't have echo cancellation hardware available on any PRI. 3. You must specifically provision the PRI with echo cancellation. I've found acceptable echo cancellation on the PRI with Asterisk's echo cancellation software on the TE405P with the following: - agressive cancellation echocancel=yes echocancelwhenbridged=yes echotraining=500 No need to worry about the echo canceller killing fax/data connections since just like the real echo cancellation hardware, asterisk will disable the echo cancel routines when it hears the correct disable tone on the line. You'll see something like zaptel Disabled echo canceller because of tone (tx) on channel 13. We were really having a lot of echo troubles but 20040831 CVS HEAD seems to have really helped, although it was certainly acceptable with 20040806 CVS HEAD. I haven't been able to locate a good hardware echo canceller on ebay yet (I keep missing the auctions). :-) -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] Question about dial out via Zap
Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1 answered SIP/6149236651-eafb Urgent handler -- Hungup 'Zap/1-1' Urgent handler Urgent handler -- Hungup 'Zap/2-1' Urgent handler Urgent handler ___ 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] Question about dial out via Zap
Found out something strange.. In zapata.conf if I change the signalling from featd to em_w I'm able to dial out without a problem. But I'm unable to get calls in because of the featd data sent. Change it back to featd and I'm now able to call in but unable to call out. So my question is do I need to do something when calling out for featd? It looks to me like a problem with featd. Below is a copy of my zapata.conf file. zapata.conf [channels] context=from-analog signalling=featd ;signalling=em_w group=1 channel = 1-12 usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes useincomingcalleridonzaptransfer=yes callerid=asreceived echocancel=yes echocancelwhenbridged=yes rxgain=1.0 txgain=1.0 musiconhold=default Thanks Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Monday, August 23, 2004 8:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Question about dial out via Zap Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1 answered SIP/6149236651-eafb Urgent handler -- Hungup 'Zap/1-1' Urgent handler Urgent handler -- Hungup 'Zap/2-1' Urgent handler Urgent handler ___ 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] Question about dial out via Zap
You probably have a mistake in your dialplan in extensions.conf and/or zapata.conf. And my mind reader failed to compile today, so I have no more guesses for you as you did not post anything about the pertiant configs or type of phones involved. Lyle - Original Message - From: Hall, Eric M. [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 23, 2004 7:32 AM Subject: [Asterisk-Users] Question about dial out via Zap Group When I dial a phone number that should go out to the telco my local phone rings. Does anyone have any hits ? Thanks Asterisk Ready. *CLI -- Called g1/6144196143 Urgent handler Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler Urgent handler -- Called 6149236651 Urgent handler -- SIP/6149236651-1d93 is ringing Urgent handler -- Zap/1-1 answered SIP/6149236651-eafb Urgent handler -- Hungup 'Zap/1-1' Urgent handler Urgent handler -- Hungup 'Zap/2-1' Urgent handler Urgent handler ___ 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] Question about TE405P
Dear all: Doesanybody know is it possible to use the board TE405P with only one port configured as follow, or I have to use the 4 ports at the same time ? Thanks, Angel ZAPTEL span=1,1,0,ccs,hdb3bchan=1-15dchan=16bchan=17-31 ZAPATA [channels]context=menu-general switchtype=euroisdn pridialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=noechocancel=yes immediate=nochannel = 1-15,17-29__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Re: [Asterisk-Users] Question about TE405P
You can use as many ports as you want. Just define how many ports (spans) you're gonna use near the beginning of /etc/zaptel.conf. span=1 Good luck Bruno Angel Diaz wrote: Dear all: Does anybody know is it possible to use the board TE405P with only one port configured as follow, or I have to use the 4 ports at the same time ? Thanks, Angel ZAPTEL span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 ZAPATA [channels] context=menu-general switchtype=euroisdn pridialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no echocancel=yes immediate=no channel = 1-15,17-29 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Question about TE405P
On Wednesday 18 August 2004 10:31, Angel Diaz wrote: Does anybody know is it possible to use the board TE405P with only one port configured as follow, or I have to use the 4 ports at the same time ? The four ports are independent of one another with one exception: they all share the same clock source (whichever port you configure as the primary source is the clocking source for the entire card). -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] Question about TE405P
Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 zapata.conf [channels] context=default switchtype=euroisdn pridialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaitingcallerid=yes language=en immediate=no channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers!
Re: [Asterisk-Users] Question about TE405P
I'll need somethine like this span=1,1,0,ccs,hdb3,crc4bchan=1-15bchan=17-31dchan=16 span=2,1,0,ccs,hdb3,crc4bchan=32-46bchan=48-62dchan=47 Setting timing source to 0 onspanscould give you some problems.. (at least I've had them) --- channel = 1-15,17-31 channel = 32-46,48-62 Kind RegardsClaus Futtrup - Original Message - From: Angel Diaz To: [EMAIL PROTECTED] Sent: Thursday, August 12, 2004 11:35 PM Subject: [Asterisk-Users] Question about TE405P Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 zapata.conf [channels] context=default switchtype=euroisdn pridialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaitingcallerid=yes language=en immediate=no channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 Do you Yahoo!?Yahoo! Mail - 50x more storage than other providers! ---Outgoing mail is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004
RE: [Asterisk-Users] Question about TE405P
No - your settings are not correct. Try something like this: ZAPTEL #E400P (or TE410P in E1 mode) setup #note, may need to add ,crc4 to end of span lines: #first quad board span=1,1,0,ccs,hdb3 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 #first quad board bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 ZAPATA: immediate=no switchtype=EuroISDN signalling=pri_net pridialplan=unknown context=incoming usecallerid=yes group=1 signalling=pri_cpe channel = 1-15,17-31 channel = 32-46,48-62 channel = 63-77,79-93 channel = 94-108,110-124 Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angel Diaz Sent: Thursday, August 12, 2004 2:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Question about TE405P Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 zapata.conf [channels] context=default switchtype=euroisdn pridialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaitingcallerid=yes language=en immediate=no channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 Do you Yahoo!? Yahoo! Mail http://us.rd.yahoo.com/mail_us/taglines/50x/*http://promotions.yahoo.com/ne w_mail/static/efficiency.html - 50x more storage than other providers! ___ 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] Question about TE405P
Sorry, remove that extraneous signalling=pri_net. You should just have the pri_cpe. -Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com No - your settings are not correct. Try something like this: ZAPTEL #E400P (or TE410P in E1 mode) setup #note, may need to add ,crc4 to end of span lines: #first quad board span=1,1,0,ccs,hdb3 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 #first quad board bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 ZAPATA: immediate=no switchtype=EuroISDN signalling=pri_net pridialplan=unknown context=incoming usecallerid=yes group=1 signalling=pri_cpe channel = 1-15,17-31 channel = 32-46,48-62 channel = 63-77,79-93 channel = 94-108,110-124 Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angel Diaz Sent: Thursday, August 12, 2004 2:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Question about TE405P Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 zapata.conf [channels] context=default switchtype=euroisdn pridialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaitingcallerid=yes language=en immediate=no channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 channel = 1-15,17-31 Do you Yahoo!? Yahoo! Mail http://us.rd.yahoo.com/mail_us/taglines/50x/*http://promotions.yahoo.com/ne w_mail/static/efficiency.html - 50x more storage than other providers! ___ 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] Question when using a Cisco as a PSTN GW
HI all I have a little question, and since there is a alot of Cisco Gurus somebody might be able to help me. I think It is an easy problem. My PSTN proviver strips the first digit in the callerid on all incoming calls. So when the call reaches my Asterisk I am missing a 0 in the CLID I guess it should be easy to prepend a digit on all incoming calls on a Cisco 5350 ? But I am unsure how a translationrule for that would look like. Anybody ? /Regards 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] question regarding Asterisk. X-Lite, and firewall
Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite (phone A) is on Win2K, with IP 192.168.1.100. They are both behind a router with dynamic IP address. Assume its public IP is aaa.bbb.ccc.ddd. I have another X_Lite SIP phone (phone B) that is NOT in the LAN I mentioned above. Rather, it has its own public IP address, say eee.fff.ggg.hhh. I have configured the router to forward all traffic to its port 5161 to Asterisk server's 5060 port, and configured SIP phone A to use 192.168.1.102:5060 and phone B aaa.bbb.ccc.ddd:5161 as proxy server respectively. Both phones registered successfully. Now, I used phone B to call phone A. The entire SIP hand-shake went through successfully. However, I can only get voice from phone A to phone B, not the other direction. I found that RTP traffic went from phone A - Asterisk - phone B. However, on the other direction, phone B tried to use 192.168.1.102 as destination of Asterisk to send voice too. Obviously, the IP is a private IP, hence, is not reachable. How do I change configuration of Asterisk so that phone B can use aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address? By the way, both directions use UDP protocol. Thanks! Michael Wang [EMAIL PROTECTED] 2004-07-20 ___ 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] question regarding Asterisk. X-Lite, and firewall
On Wed, 21 Jul 2004, Michael Wang wrote: How do I change configuration of Asterisk so that phone B can use aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address? sounds like * is using reinvite to get itself out of the loop and let the phones send RTP directly between themselves. Because of the NAT, this won't work. To prevent * from sending the reinvite, and to keep RTP traffic flowing through *, try using nat=yes and/or canreinvite=no in sip.conf (you choose which section, general or phone-specific) Greg ___ 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] question regarding Asterisk. X-Lite, and firewall
Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite (phone A) is on Win2K, with IP 192.168.1.100. They are both behind a router with dynamic IP address. Assume its public IP is aaa.bbb.ccc.ddd. I have another X_Lite SIP phone (phone B) that is NOT in the LAN I mentioned above. Rather, it has its own public IP address, say eee.fff.ggg.hhh. I have configured the router to forward all traffic to its port 5161 to Asterisk server's 5060 port, and configured SIP phone A to use 192.168.1.102:5060 and phone B aaa.bbb.ccc.ddd:5161 as proxy server respectively. Both phones registered successfully. Now, I used phone B to call phone A. The entire SIP hand-shake went through successfully. However, I can only get voice from phone A to phone B, not the other direction. I found that RTP traffic went from phone A - Asterisk - phone B. However, on the other direction, phone B tried to use 192.168.1.102 as destination of Asterisk to send voice too. Obviously, the IP is a private IP, hence, is not reachable. How do I change configuration of Asterisk so that phone B can use aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address? By the way, both directions use UDP protocol. Thanks! Michael Wang [EMAIL PROTECTED] 2004-07-20
Re: [Asterisk-Users] question regarding Asterisk. X-Lite, and firewall
Michael Wang wrote: Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite (phone A) is on Win2K, with IP 192.168.1.100. They are both behind a router with dynamic IP address. Assume its public IP is aaa.bbb.ccc.ddd. I have another X_Lite SIP phone (phone B) that is NOT in the LAN I mentioned above. Rather, it has its own public IP address, say eee.fff.ggg.hhh. I have configured the router to forward all traffic to its port 5161 to Asterisk server's 5060 port, and configured SIP phone A to use 192.168.1.102:5060 and phone B aaa.bbb.ccc.ddd:5161 as proxy server respectively. Both phones registered successfully. Now, I used phone B to call phone A. The entire SIP hand-shake went through successfully. However, I can only get voice from phone A to phone B, not the other direction. I found that RTP traffic went from phone A - Asterisk - phone B. However, on the other direction, phone B tried to use 192.168.1.102 as destination of Asterisk to send voice too. Obviously, the IP is a private IP, hence, is not reachable. try this in your sip.conf disallow=all allow=ulaw allow=alaw nat=yes or use a STUN server 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
[Asterisk-Users] Question about Asterisk Installation
Excuse for my bad English, but I have a problem after the compilation of Asterisk-addons: when I try to execute the simbolic link ln -s /usr/lib/mysql/libmysqlclient.so /usr/lib/libmysqlclient.so this produce an error because the source object libmysqlclient.so is not present in any directory. So I can't continue with Asterisk compilation. May help me? thanks Vincenzo ___ 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] Question about Cisco IP Phone 7960
I know this is a little off list but I can't think of a better place to ask this question. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf Any ideas? Again sorry this is off topic ___ 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] Question about Cisco IP Phone 7960
On 08/07/2004 at 08:21 Hall, Eric M. wrote: I know this is a little off list but I can't think of a better place to ask this question. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf Any ideas? Again sorry this is off topic Make sure you changed ALL the configs to point to the CORRECT image file.. OS79XX.TXT should contain P0S3-07-1-00 and your SIPmac or SIPDefault.cnf should contain image_version: P0S3-07-1-00 iirc the default in OS79XX.TXT is the unsigned image... HTH Andy ___ 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] Question about Cisco IP Phone 7960
You the MAN!!! I drop the P of the P0S3-07-1-00 Everything is golden now! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Powell Sent: Thursday, July 08, 2004 9:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Question about Cisco IP Phone 7960 On 08/07/2004 at 08:21 Hall, Eric M. wrote: I know this is a little off list but I can't think of a better place to ask this question. I upgrade the phone to 7.1 and it installed the Universal Application Loader. Now I'm getting Protocol Application Invalid after it reads tftp SIP(MAC).cnf Any ideas? Again sorry this is off topic Make sure you changed ALL the configs to point to the CORRECT image file.. OS79XX.TXT should contain P0S3-07-1-00 and your SIPmac or SIPDefault.cnf should contain image_version: P0S3-07-1-00 iirc the default in OS79XX.TXT is the unsigned image... HTH Andy ___ 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] Question about x100P and zap
I have2 X100P card and configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt I changed the area codes to match mine. When I try to dial out I get app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' A zap show channels gives me this Chan Extension Context Language MusicOnHold 1 from-analog en Any ideas?
RE: [Asterisk-Users] Question about x100P and zap
Your $EXTENs need to be changed to
${EXTEN}. Youll also need to include any substr #s within
the brackets (ie. ${EXTEN:1}).
-wade
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Hall, Eric M.
Sent: Monday, July 05, 2004 10:25
AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Question
about x100P and zap
I have2 X100P card and configured everything based on
configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt
I changed the area codes to match mine.
When I try to dial out I get
app_dial.c:554 dial_exec: Unable to create channel of type
'Zap'
A zap show channels gives me this
Chan Extension
Context Language
MusicOnHold
1
from-analog
en
Any ideas?
RE: [Asterisk-Users] Question about x100P and zap
I did as you stated however I get the same error. Here is
my config file. Did I miss something?
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wade J.
WepplerSent: Monday, July 05, 2004 10:54 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Question
about x100P and zap
Your $EXTENs need to
be changed to ${EXTEN}. Youll also need to include any substr #s within
the brackets (ie. ${EXTEN:1}).
-wade
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.Sent: Monday, July 05, 2004 10:25
AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Question about
x100P and zap
I have2 X100P card and
configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt
I changed the area codes to match
mine.
When I try to dial out I get
app_dial.c:554 dial_exec: Unable to
create channel of type 'Zap'
A zap show channels gives me
this
Chan Extension
Context Language
MusicOnHold
1
from-analog
en
Any
ideas?
[general]
static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
;
;
;
[from-sip-external]
;
; Take unknown callers that are sending calls to our system,
; and send them to the appropriate extension. It is in this
; area that we do name-to-number mapping for SIP extensions.
;
; This context will allow calls to [EMAIL PROTECTED] or calls
; to [EMAIL PROTECTED] to be answered on the relevant SIP
; phones. We also do some name-to-number mapping here; see below.
;
; The SIP URI of sip://[EMAIL PROTECTED] will be handled here.
; Note that we assume Sidney is on the SIP phone described as
; extension 2000 in sip.conf, so this short routine just
; re-directs the call flow recursively back into the same
; context, but we change the extension and priority. Since
; we're including [local-extensions], this will get picked up
; by the dialplan contained in local-extensions.
;
; I could be more space-efficient and put all these lines into
; a single regexp, but for clarity I put them each on their
; own lines.
;
; Here are Sidney's aliases
;
exten = sidney,1,Goto(2000,1)
exten = sidney.zweibel,1,Goto(2000,1)
exten = info,1,Goto(2000,1)
;
; ...and John's aliases.
;
exten = john,1,Goto(2001,1)
exten = john.whorfin,1,Goto(2001,1)
exten = sales,1,Goto(2001,1)
;
;
; Include the numbers which we have defined in local-extensions
; and allow them to be accessed from within this context. This
; is how we are able to use the Goto commands above, since
; we will be including extensions 2000 and 2001 (and 0 and 2999)
; as available extensions to which we may re-route calls within
; this context.
;
include = local-extensions
;
; If the line hangs up, it's always good to have the h
; extension in each context that is the master handler
; for calls. This cleanly exits and closes dial path routes.
;
exten = h,1,Hangup
;
; The user has dialed an invalid number, which means that
; there was no match by any other matching routines. Set an
; absolute timeout on the call (15 seconds), play a Congestion
; tone, and hangup. We set the absolute timeout to prevent easy
; DoS attacks from consuming too much bandwidth. However, it
; is possible that we could still be attacked in some fashion
; by someone making many calls to bogus numbers on our server.
; We could reduce this threat by removing the Congestion
; playback and going straight to hangup, but that is very
; difficult to debug at the remote end, so we are good VoIP
; citizens and we create some audio if the call reaches us.
;
exten = i,1,AbsoluteTimeout(15)
exten = i,2,Congestion
exten = i,3,Hangup
;
;
;
[from-sip-internal]
; Calls that come in from our two SIP phones will land here
; first and match against extensions listed below.
;
; The context [from-sip-internal] is really just a collection
; of include statements that pull the extension matching lists
; in from other contexts. A well-designed dialplan segregates
; extensions with similar functions into contexts, and then
; uses the include referencer. This should be a familiar
; concept to anyone who does programming - segmenting a block
; of phone numbers makes them more re-usable in a generic way
; so that the administrator can avoid re-typing the same configs
; over and over.
;
; First, we include [local-extensions], since that's what
; we should try matching on first. If anyone on one of our
; local SIP phones dials an extension that appears in
; [local-extensions], then send the call to whatever priority
; list exists for that number. This is for local-to-local call
; termination.
;
include = local-extensions
;
; Next, we include and try to match against extensions contained
; in [always-out-pots]. These are mostly wildcarded matches,
; so we make sure to put them second when we list our include
; statements, since Asterisk will process the matching
RE: [Asterisk-Users] Question about x100P and zap
Ruuing * in debug I get this
*CLI Jul 5 11:21:02 NOTICE[-1221170256]:
app_dial.c:554 dial_exec: Unable to create channel of type 'Zap' ==
Everyone is busy at this time
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wade J.
WepplerSent: Monday, July 05, 2004 10:54 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Question
about x100P and zap
Your $EXTENs need to
be changed to ${EXTEN}. Youll also need to include any substr #s within
the brackets (ie. ${EXTEN:1}).
-wade
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.Sent: Monday, July 05, 2004 10:25
AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Question about
x100P and zap
I have2 X100P card and
configured everything based on configs here http://www.onlamp.com/onlamp/2004/01/22/examples/config_files.txt
I changed the area codes to match
mine.
When I try to dial out I get
app_dial.c:554 dial_exec: Unable to
create channel of type 'Zap'
A zap show channels gives me
this
Chan Extension
Context Language
MusicOnHold
1
from-analog
en
Any
ideas?
RE: [Asterisk-Users] Question about x100P and zap
You might want to check to make sure your signaling is correct in Zapata.conf and zaptel.conf. They should be fxs_ks and fxsks respectively. If you run zttool from your zaptel source directly, does it tell you that you have an active and green/OK alarm X100P? -wade ___ 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] Question - TDM40B - Hunt Group Possibility??
Title: Message I was wondering if this is possible. I have a situation where I am connecting to a third party voicemail system from asterisk. I know this does not make since to everyone, but it has to be this way. Basically - I have an application that runs on the Asterisk system and when an employee calls into this system, they have an option to check there voicemail. This is where it needs to go over to the voicemail system. I would usually use an FXO card for this, but the other phone vendor I am working with is wondering is it possible to put the FXS cards I have in a hunt group - then I could call one of these ports and would ring the other voicemail system. If this can't be done that's fine - I have some FXO cards on order... Just thought I would check if anyone has ever done anything like this before. Thanks, Geoff Clark
[Asterisk-Users] Question IAX and SIP bound to different IP's on the same * box
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, May 25, 2004 5:30 AM
To: [EMAIL PROTECTED]
Subject: Asterisk-Users digest, Vol 1 #3891 - 8 msgs
Send Asterisk-Users mailing list submissions to
[EMAIL PROTECTED]
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
[EMAIL PROTECTED]
You can reach the person managing the list at
[EMAIL PROTECTED]
When replying, please edit your Subject line so it is more specific
than Re: Contents of Asterisk-Users digest...
Today's Topics:
1. RE: Newbie extensions.conf I need to include [SMS] context. (Gary
Ruddock)
2. Re: Document - contains malware (Trevor Peirce)
3. RE: Newbie extensions.conf I need to include [SMS] context. (Jay
Milk)
4. Re: Sip Registration Problem (Olle E. Johansson)
5. Using Ser and Asterisk together (=?iso-8859-1?q?Aiden=20Chew?=)
6. RE: 100 analog phones?? HOWTO? ([EMAIL PROTECTED])
7. SipTone II and Choppy/Stuttering Audio (Nick Grindley)
8. RE: Meetme Options (new one) (Ben Merrills)
--__--__--
Message: 1
From: Gary Ruddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Newbie extensions.conf I need to include
[SMS] context.
Date: Tue, 25 May 2004 07:22:29 +0100
Reply-To: [EMAIL PROTECTED]
I have been up all night and I gotta go to bed.
If there's anyone out there using asterisk to send SMS text messages in
the
UK with BT please gis a clue. Do I need to get the latest asterisk CVS?
Could anyone be so kind as to tell me how to modify this dialplan to
accept
and send SMS text messages. Do I need to update my basic Asterisk to
include SMS functionality? In the example contexts a reference is made
to
/usr/lib/asterisk/smsin and I can't find that file.
I know that [local] is executed first and it includes other contexts. I
need to add these two contexts
[smsdial] ; create and send a text message, expects
number+message
and
connect to 17094009
exten = _X.,1,SMS(${CALLERIDNUM},,${EXTEN},${CALLERIDNAME})
exten = _X.,2,SMS(${CALLERIDNUM})
exten = _X.,3,Hangup
and
[incoming]
exten = _XX/_8005875290,1,SMS(${EXTEN:3},a)
exten = _XX/_8005875290,2,System(/usr/lib/asterisk/smsin
${EXTEN:3})
exten = _XX/_80058752[0-8]0,1,SMS(${EXTEN:3}${CALLERIDNUM:8:1},a)
exten = _XX/_80058752[0-8]0,2,System(/usr/lib/asterisk/smsin
${EXTEN:3}${CALLERIDNUM:8:1})
exten = _XX/_80058752X0,3,Hangup
*** my extensions.conf ***
[general]
static=yes
writeprotect=no
[globals]
TRUNK=Zap/g1; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[trunkint]
;exten = _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
;exten = _9011.,2,Congestion
[trunkld]
exten = _90XXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _90XXXNXX,2,Congestion
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Congestion
exten = _907N,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _907N,2,Congestion
[trunktollfree]
exten = _90800NX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _90800NX,2,Congestion
[international]
ignorepat = 9
include = longdistance
include = trunkint
[longdistance]
;ignorepat = 9
;include = local
include = trunkld
[local]
ignorepat = 9
;include = default
include = parkedcalls
include = trunklocal
include = trunktollfree
include = trunkld
exten = 6001,1,Dial(SIP/6001,20,tr)
exten = 6002,1,Dial(SIP/6002,20,tr)
exten = 07,1,Answer
exten = 07,2,wait(2)
exten = 07,3,playback(welcome)
exten = 07,4,GotoIf($[foo${CALLERIDNUM} = foo]?7:5)
exten =
07,5,Queue(salesq,,https://swiftdrinks.com/admin/callerid.php?calle
rid=${CALLERIDNUM})
exten = 07,6,Hangup
exten = 07,7,Wait(2)
exten = 07,8,Playback(privacy-unident)
exten = 07,9,Hangup
exten = 2500,1,Dial(Zap/32,40)
exten = 2500,2,VoiceMail2(u2500)
exten = 2500,3,Hangup
exten = 2500,102,VoiceMail2(b2500)
exten = 2500,103,Hangup
exten = 2501,1,Dial(Zap/33,40)
exten = 2501,2,VoiceMail2(u2500)
exten = 2501,3,Hangup
exten = 2501,102,VoiceMail2(b2501)
exten = 2501,103,Hangup
exten = 81,1,AddQueueMember(salesq|Zap/32)
exten = 81,2,wait(1)
exten = 81,3,Playback(agent-loginok)
exten = 81,4,wait(1)
exten = 81,5,Hangup
exten = 82,1,RemoveQueueMember(salesq|Zap/32)
exten = 82,2,wait(1)
exten = 82,3,Playback(agent-loggedoff)
exten = 82,4,wait(1)
exten = 82,5,Hangup
exten = 95,3,Playback(agent-loginok)
exten = 95,4,wait(1)
exten = 95,5,Hangup
exten = 96,1,RemoveQueueMember(salesq|SIP/6001)
exten = 96,2,wait(1)
exten = 96,3,Playback(agent-loggedoff)
exten = 96,4,wait(1)
exten = 96,5,Hangup
exten = 97,1,AddQueueMember(salesq|SIP/6002)
exten = 97,2,wait(1)
exten = 97,3,Playback(agent-loginok)
exten =
[Asterisk-Users] Question about Asterisk and its use
Hello I am very new in this area, just start reading about Asterisk and VoIP 2 days ago. I am very interested in this product but not really getting good information on. Will appreciate of someone an answer these question in detail or direct me to right documents: 1. How to setup and use this Product? 2. If i have Asterisk installed on one Linux server remortly, Can i use that Asterisk server for my call routhing using soft phone installed on my desktop. Thanks Deepak This message was sent using IMP, the Internet Messaging Program. ___ 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] Question about Asterisk and its use
Question number one is overwhelming... How do you want to use this product. That will dictate how you set it up. In any case, there is extensive documentation on this, please go there first and formulate more specific questions that we can help you with. Try here for starts: http://www.automated.it/guidetoasterisk.htm [EMAIL PROTECTED] wrote: Hello I am very new in this area, just start reading about Asterisk and VoIP 2 days ago. I am very interested in this product but not really getting good information on. Will appreciate of someone an answer these question in detail or direct me to right documents: 1. How to setup and use this Product? 2. If i have Asterisk installed on one Linux server remortly, Can i use that Asterisk server for my call routhing using soft phone installed on my desktop. Thanks Deepak This message was sent using IMP, the Internet Messaging Program. ___ 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] Question about Asterisk and its use
Perhaps consult an Asterisk consultant? I hear these guys are pretty good at helping and can point you in the right direction for a small fee ;-) Otherwise visit the wiki at www.voip-info.org and expect to spend a few weeks at a minimum sorting through it all like the rest of us did. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 10, 2004 11:22 AM Subject: [Asterisk-Users] Question about Asterisk and its use Hello I am very new in this area, just start reading about Asterisk and VoIP 2 days ago. I am very interested in this product but not really getting good information on. Will appreciate of someone an answer these question in detail or direct me to right documents: 1. How to setup and use this Product? 2. If i have Asterisk installed on one Linux server remortly, Can i use that Asterisk server for my call routhing using soft phone installed on my desktop. Thanks Deepak This message was sent using IMP, the Internet Messaging Program. ___ 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] Question about Asterisk and its use
Perhaps point him to that beta2 users manual .pdf as well, its helped me greatly after some kind soul sent it to me when I was where he is now.. (I dont have the url handy, could someone cough it up please?) IMHO, its a very good place to get the overview, and then the wiki makes a lot more sense. At 15:25 5/10/2004, you wrote: Perhaps consult an Asterisk consultant? I hear these guys are pretty good at helping and can point you in the right direction for a small fee ;-) Otherwise visit the wiki at www.voip-info.org and expect to spend a few weeks at a minimum sorting through it all like the rest of us did. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 10, 2004 11:22 AM Subject: [Asterisk-Users] Question about Asterisk and its use Hello I am very new in this area, just start reading about Asterisk and VoIP 2 days ago. I am very interested in this product but not really getting good information on. Will appreciate of someone an answer these question in detail or direct me to right documents: 1. How to setup and use this Product? 2. If i have Asterisk installed on one Linux server remortly, Can i use that Asterisk server for my call routhing using soft phone installed on my desktop. Thanks Deepak This message was sent using IMP, the Internet Messaging Program. ___ 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
[Asterisk-Users] Question of Asterisk timer to get Conference work
Hello, Can someone help me. I got zaptel.0.9.1.tar.gz from ftp.asterisk.org, And then I uncomment the line with MODULES # ztdummy, run make clean, make, make install But errors happens as follows: -- make: zaptel.c:5937: storage size of `zt_fops' isn't known/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never definedmake: *** [zaptel.o] Error 1 -- make install: zaptel.c:5937: storage size of `zt_fops' isn't known/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never definedmake: *** [zaptel.o] Error 1 Is there anybody ever install this timer driver, please tell me what's wrong? Thanks! Chunghwa Telecom BTA Tech. LabE-mail:[EMAIL PROTECTED]
[Asterisk-Users] Question of Asterisk timer to get Conference work
Hello, Can someone help me. I got zaptel.0.9.1.tar.gz from ftp.asterisk.org, And then I uncomment the line with MODULES # ztdummy, run make clean, make, make install But errors happens as follows: -- make: zaptel.c:5937: storage size of `zt_fops' isn't known/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never definedmake: *** [zaptel.o] Error 1 -- make install: zaptel.c:5937: storage size of `zt_fops' isn't known/usr/include/linux/proc_fs.h:193: warning: `create_proc_read_entry' declared `static' but never definedmake: *** [zaptel.o] Error 1 Is there anybody ever install this timer driver, please tell me what's wrong? Thanks! Chunghwa Telecom BTA Tech. LabE-mail:[EMAIL PROTECTED]
[Asterisk-Users] Question about prepaid db
Hello, Somebody has an example with all data loaded in the base for prepaid? or an example of a base that this working?... Thanks... Julio
Re: [Asterisk-Users] Question about prepaid db
Somebody made run prepaid?... - Original Message - From: Julio To: [EMAIL PROTECTED] Sent: Monday, April 19, 2004 2:38 PM Subject: [Asterisk-Users] Question about prepaid db Hello, Somebody has an example with all data loaded in the base for prepaid? or an example of a base that this working?... Thanks... Julio
[Asterisk-Users] Question receiving calls via SIP
Hello-
I am in the process of adding a new provider to my asterisk box (both
for outbound termination as well as inbound DID). They are going to be
delivering and receiving traffic via SIP only.
Now, in IAX via Voicepulse or others I know that I can simply have one
registration statement along with an inbound context, then in
extension.conf map the outbound context.
from iax.conf:
register = in-:[EMAIL PROTECTED]
from extensions.conf
[voicepulse-in]
exten = 212xxx,1,Dial(${PHONES1}${PHONES2},30)
exten = 212xxx,2,Voicemail2(u${PHONES1VM})
exten = 212xxx,3,Hangup
I know this way I only have to register once, but can receive calls on
several inbound DID numbers without any problem, provided they are all
mapped similar to what I have above within extensions.conf.
My question is whether or not the same thing will work for a sip
provider, as it will be pretty cumbersome from a networking standpoint
to have a registration statement for each DID (as opposed to simply
having a new extension statement). In the syntax of the sip
registration statement it appears to always end with the /extension
that is supposed to be associated with that account.
Can the /extension be left off of the statement in sip.conf, and picked
up in the same way DID's are used in iax.conf? If not, why are there
separate methods for registering each protocol, would seem cleaner to
have a consistent way of dealing with this.
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] question about CPU usage
try to do ps -auxm to list all the threads of the asterisk. Then connect with gdb to the thread that takes 99% of CPU and find out what it's doing. Martin On Mon, 22 Mar 2004, Bill Hamlin wrote: Nope same problem. I just started it and did a couple of ps aux's and got this output: [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 91.6 1.3 115880 6676 ? R15:43 1:10 asterisk -vgcd root 20221 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.3 1.3 115880 6676 ? R15:43 1:13 asterisk -vgcd root 20223 0.0 0.1 3568 624 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 91.7 1.3 115880 6676 ? R15:43 1:16 asterisk -vgcd root 20225 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.4 1.3 115880 6676 ? R15:43 1:18 asterisk -vgcd root 20227 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.6 1.3 115880 6676 ? R15:43 1:20 asterisk -vgcd root 20229 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Monday, March 22, 2004 4:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] question about CPU usage I think Steve is referring to the following line: export LD_ASSUME_KERNEL=2.4.1 If you put this in your command line before starting asterisk, you will get around the RH9 problem of leaving zombies when AGI processes quit. Other than that, I don't think it influences CPU load. Note that the line is not necessary for Fedora Core 1 regards Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Hamlin Sent: Monday, March 22, 2004 9:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] question about CPU usage What is it about asterisk that makes this happen? My other apps that wait on a select take hardly any CPU time at all. I didn't find anything like ldassume using google. Can you tell me more about that? Thanks, Bill. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, March 22, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] question about CPU usage On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Do a google search. I believe there is a export line you need for RH to behave more sanely. Something like ldassume_2_4_1. Or you could switch to a more free distro and it will fix itself. -- 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 ___ 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 ___ 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] question about CPU usage
Back in February I found * pinning the CPU (Slackware 9.1, Feb CVS). I
ran a strace and found that it was looping on this:
-begin-
write(1, \nUse STOP NOW to shutdown Asteri..., 35) = -1 EIO
(Input/output erro
r)
write(1, *CLI , 6) = -1 EIO (Input/output error)
read(0, , 1) = 0
ioctl(0, SNDCTL_TMR_STOP, {B38400 opost isig icanon echo ...}) = -1 EIO
(Input/o
utput error)
-end-
Mark kindly responded to me:
-begin-
In the mean time try running asterisk with no console. This is bug #864.
Preliminary analysis shows that after a restart now, one of the
ioctl()'s performed by editline fails with -1. Ignoring the ioctl made
the CLI non-functional. Happy to get any help I can in this regard.
-end-
Hope this helps.
Cheers
Martin Pycko wrote:
try to do ps -auxm to list all the threads of the asterisk.
Then connect with gdb to the thread that takes 99% of CPU and find out
what it's doing.
Martin
On Mon, 22 Mar 2004, Bill Hamlin wrote:
Nope same problem. I just started it and did a couple of ps aux's and got
this output:
[EMAIL PROTECTED] root]# ps aux|grep ast
root 20140 91.6 1.3 115880 6676 ? R15:43 1:10
asterisk -vgcd
___
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] question about CPU usage
I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Thanks, Bill Hamlin. ___ 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] question about CPU usage
On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Do a google search. I believe there is a export line you need for RH to behave more sanely. Something like ldassume_2_4_1. Or you could switch to a more free distro and it will fix itself. -- 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] question about CPU usage
What is it about asterisk that makes this happen? My other apps that wait on a select take hardly any CPU time at all. I didn't find anything like ldassume using google. Can you tell me more about that? Thanks, Bill. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, March 22, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] question about CPU usage On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Do a google search. I believe there is a export line you need for RH to behave more sanely. Something like ldassume_2_4_1. Or you could switch to a more free distro and it will fix itself. -- 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 ___ 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] question about CPU usage
I think Steve is referring to the following line: export LD_ASSUME_KERNEL=2.4.1 If you put this in your command line before starting asterisk, you will get around the RH9 problem of leaving zombies when AGI processes quit. Other than that, I don't think it influences CPU load. Note that the line is not necessary for Fedora Core 1 regards Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Hamlin Sent: Monday, March 22, 2004 9:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] question about CPU usage What is it about asterisk that makes this happen? My other apps that wait on a select take hardly any CPU time at all. I didn't find anything like ldassume using google. Can you tell me more about that? Thanks, Bill. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, March 22, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] question about CPU usage On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Do a google search. I believe there is a export line you need for RH to behave more sanely. Something like ldassume_2_4_1. Or you could switch to a more free distro and it will fix itself. -- 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 ___ 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] question about CPU usage
Bill, I think your looking for setting the environment variable LD_ASSUME_KERNEL=2.4.1. If I remember correctly this effectively disables the new NTPL (new threading model) in RH9. Hope this helps. Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Hamlin Sent: Monday, March 22, 2004 1:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] question about CPU usage What is it about asterisk that makes this happen? My other apps that wait on a select take hardly any CPU time at all. I didn't find anything like ldassume using google. Can you tell me more about that? Thanks, Bill. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, March 22, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] question about CPU usage On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Do a google search. I believe there is a export line you need for RH to behave more sanely. Something like ldassume_2_4_1. Or you could switch to a more free distro and it will fix itself. -- 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 ___ 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] question about CPU usage
export LD_ASSUME_KERNEL=2.4.1 - Original Message - From: Bill Hamlin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 22, 2004 4:22 PM Subject: RE: [Asterisk-Users] question about CPU usage What is it about asterisk that makes this happen? My other apps that wait on a select take hardly any CPU time at all. I didn't find anything like ldassume using google. Can you tell me more about that? Thanks, Bill. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, March 22, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] question about CPU usage On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Do a google search. I believe there is a export line you need for RH to behave more sanely. Something like ldassume_2_4_1. Or you could switch to a more free distro and it will fix itself. -- 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 ___ 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] question about CPU usage
On Mon, 2004-03-22 at 15:22, Bill Hamlin wrote: I didn't find anything like ldassume using google. Can you tell me more about that? It's in the RedHat 9 RELEASE NOTES. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] question about CPU usage
Nope same problem. I just started it and did a couple of ps aux's and got this output: [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 91.6 1.3 115880 6676 ? R15:43 1:10 asterisk -vgcd root 20221 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.3 1.3 115880 6676 ? R15:43 1:13 asterisk -vgcd root 20223 0.0 0.1 3568 624 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 91.7 1.3 115880 6676 ? R15:43 1:16 asterisk -vgcd root 20225 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.4 1.3 115880 6676 ? R15:43 1:18 asterisk -vgcd root 20227 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.6 1.3 115880 6676 ? R15:43 1:20 asterisk -vgcd root 20229 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Monday, March 22, 2004 4:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] question about CPU usage I think Steve is referring to the following line: export LD_ASSUME_KERNEL=2.4.1 If you put this in your command line before starting asterisk, you will get around the RH9 problem of leaving zombies when AGI processes quit. Other than that, I don't think it influences CPU load. Note that the line is not necessary for Fedora Core 1 regards Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Hamlin Sent: Monday, March 22, 2004 9:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] question about CPU usage What is it about asterisk that makes this happen? My other apps that wait on a select take hardly any CPU time at all. I didn't find anything like ldassume using google. Can you tell me more about that? Thanks, Bill. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, March 22, 2004 4:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] question about CPU usage On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? Do a google search. I believe there is a export line you need for RH to behave more sanely. Something like ldassume_2_4_1. Or you could switch to a more free distro and it will fix itself. -- 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 ___ 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] question about CPU usage
On Mon, Mar 22, 2004 at 03:49:29PM -0500, Bill Hamlin wrote: I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling sleep(0) or something simlar so as to relinquish the machine but otherwise polling like crazy? http://lists.digium.com/pipermail/asterisk-dev/2003-December/002391.html -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ 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] Question regarding MusicOnHold ...
I've been told that MusicOnHold is *incredibly* picky about the mp3s that it plays. I've experimented with the sample and a host of other constant bitrate mp3s, and even some VBR ones, and I can't get any sort of consistent workability. Even the sample doesn't work but maybe 10% of the time. I'm using mpg123 version 0.59r. Does anyone have any advice they can give me on getting MusicOnHold to work the majority of the time? Or... if I need to provide any additional information, let me know. Thanks! -- Daniel Prather CityNet, LLC [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part
[Asterisk-Users] Question about 'zap show channels'
Does anybody get any data in the 'Extension' column of the 'zap show channels' output? I'm at a loss as to where it would be getting any information to populate this column. I've looked in the sample zapata.conf chan_zap.c. I've tried specifying extension=blah or exten=blah in zapata.conf. Nothing working incorrectly, but I'm wondering if I'm missing out on something if my Zap channels aren't set up right... Thanx, Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. 666,000,000,000 -- The number of the gigabeast. ___ 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] question for oh323 users
Thanks very much Michael. It worked but only if I configure my cisco to use g711alaw. If I config my cisco to use default g729r8 it created the below Feb 9 15:37:59 WARNING[32788]: channel.c:1856 ast_channel_make_compatible: No path to translate from H323:9242(256) to H323:28967(8) Feb 9 15:37:59 WARNING[32788]: channel.c:2245 ast_channel_bridge: Can't make H323:9242 and H323:28967 compatible Feb 9 15:37:59 WARNING[32788]: res_parking.c:226 ast_bridge_call: Bridge failed on channels H323:9242 and H323:28967 Is it because I do not have the codec g729r8 in /usr/lib/asterik/modules ? I have format_g729.so. If thats the case where could I get the codec from?? oh323.conf [codecs] codec=G729 frames=2 codec=G711A frames=20 ;codec=GSM0610 ;frames=4 codec=G7231 frames=2 Thanks for your input. Regards, Anthony ___ 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] question for oh323 users
Anthony Law wrote: Hi Gus, Thanks for your reply. I have tried below and still didn't work. exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED] or exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED] and now asterisk gives out below error Feb 6 16:12:41 WARNING[30740]: pbx.c:1773 ast_pbx_run: Channel 'H323:8915' sent into invalid extension 's' in context 'default', but no invalid handler You must define a context for the incoming calls (section [register] in oh323.conf). You need someting like this: [register] context=demo gwprefix=1905 here is exactly what I have in extension.conf [general] static=yes writeprotect=no [default] include = demo [demo] exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED] Any idea? Regards, Anthony Michael. ___ 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] question for oh323 users
Hi, Thanks for your reply. I am definite that my h323 is running on ciscoB because the below scenario is working fine. pstnciscoA-ciscoBpstn I have also eliminated access-list problem because if my access-list is applied I could see packets hiting my access-list permit tcp host 192.168.1.2 any eq 1720 (60 matches) Is my syntax below correct ?? exten = _1905XXX,1,Dial,OH323/192.168.1.3 Any help would be appreciated. Regards, Anthony - Original Message - From: Tomica Crnek [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 3:03 AM Subject: RE: [Asterisk-Users] question for oh323 users Hi, it seams to me that h.323 service on your cisco B could be down. You see packets coming to this box, but did you activate h.323. Try telnet 192.168.1.3 1720 to see if it is running. If it is, then check to see if you are allowing connections to it from 192.168.1.2 Tomica -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law Sent: Thursday, February 05, 2004 10:41 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] question for oh323 users Hi, I am trying to forward calls from one cisco gateway to another cisco gateway using asterisk cisco(5300)A 192.168.1.1 asterisk 192.168.1.2 cisco(5300)B 192.168.1.3 pstn --ciscoA-asterisk --ciscoB--pstn I have the below in my extension.conf [default] exten = _1905XXX,1,Dial,OH323/192.168.1.3 I keep getting error and I don't know what is wrong. I am able to see in my ciscoB accesslist, tcp packets are coming from 192.168.1.2 I get below error in my asterisk CLI Feb 5 16:17:01 ERROR[29716]: chan_oh323.c:1004 oh323_call: H323:0: Could not call 192.168.1.3. Feb 5 16:17:11 WARNING[29716]: pbx.c:1829 ast_pbx_run: Timeout, but no rule 't' in context 'default' It would be much appreciated if someone could point out what I am doing wrong or to any documentations. Many thanks. Regards, Anthony ___ 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] question for oh323 users
It must be: exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED] or exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED] Hope this helps, Gus - Original Message - From: Anthony Law [EMAIL PROTECTED] To: Mailing List Asterisk [EMAIL PROTECTED] Sent: Friday, February 06, 2004 11:56 AM Subject: Re: [Asterisk-Users] question for oh323 users Hi, Thanks for your reply. I am definite that my h323 is running on ciscoB because the below scenario is working fine. pstnciscoA-ciscoBpstn I have also eliminated access-list problem because if my access-list is applied I could see packets hiting my access-list permit tcp host 192.168.1.2 any eq 1720 (60 matches) Is my syntax below correct ?? exten = _1905XXX,1,Dial,OH323/192.168.1.3 Any help would be appreciated. Regards, Anthony - Original Message - From: Tomica Crnek [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, February 06, 2004 3:03 AM Subject: RE: [Asterisk-Users] question for oh323 users Hi, it seams to me that h.323 service on your cisco B could be down. You see packets coming to this box, but did you activate h.323. Try telnet 192.168.1.3 1720 to see if it is running. If it is, then check to see if you are allowing connections to it from 192.168.1.2 Tomica -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law Sent: Thursday, February 05, 2004 10:41 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] question for oh323 users Hi, I am trying to forward calls from one cisco gateway to another cisco gateway using asterisk cisco(5300)A 192.168.1.1 asterisk 192.168.1.2 cisco(5300)B 192.168.1.3 pstn --ciscoA-asterisk --ciscoB--pstn I have the below in my extension.conf [default] exten = _1905XXX,1,Dial,OH323/192.168.1.3 I keep getting error and I don't know what is wrong. I am able to see in my ciscoB accesslist, tcp packets are coming from 192.168.1.2 I get below error in my asterisk CLI Feb 5 16:17:01 ERROR[29716]: chan_oh323.c:1004 oh323_call: H323:0: Could not call 192.168.1.3. Feb 5 16:17:11 WARNING[29716]: pbx.c:1829 ast_pbx_run: Timeout, but no rule 't' in context 'default' It would be much appreciated if someone could point out what I am doing wrong or to any documentations. Many thanks. Regards, Anthony ___ 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] question for oh323 users
Hi Gus, Thanks for your reply. I have tried below and still didn't work. exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED] or exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED] and now asterisk gives out below error Feb 6 16:12:41 WARNING[30740]: pbx.c:1773 ast_pbx_run: Channel 'H323:8915' sent into invalid extension 's' in context 'default', but no invalid handler here is exactly what I have in extension.conf [general] static=yes writeprotect=no [default] include = demo [demo] exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED] Any idea? Regards, Anthony ___ 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] question for oh323 users
Hi, I am trying to forward calls from one cisco gateway to another cisco gateway using asterisk cisco(5300)A 192.168.1.1 asterisk 192.168.1.2 cisco(5300)B 192.168.1.3 pstn --ciscoA-asterisk --ciscoB--pstn I have the below in my extension.conf [default] exten = _1905XXX,1,Dial,OH323/192.168.1.3 I keep getting error and I don't know what is wrong. I am able to see in my ciscoB accesslist, tcp packets are coming from 192.168.1.2 I get below error in my asterisk CLI Feb 5 16:17:01 ERROR[29716]: chan_oh323.c:1004 oh323_call: H323:0: Could not call 192.168.1.3. Feb 5 16:17:11 WARNING[29716]: pbx.c:1829 ast_pbx_run: Timeout, but no rule 't' in context 'default' It would be much appreciated if someone could point out what I am doing wrong or to any documentations. Many thanks. Regards, Anthony ___ 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] Question on setting up asterisk with hunting lines
*My apologies if this message is posted 3 times, I was trying to sent it to the list once before I am a list-member, the second time before I was approved. Can anyone point me to some resources on using hunting lines with Asterisk? Sales support of my telco have no idea what I am trying to do. They asked what pbx system I am using, I was like Aster... never mind =) I am trying to setup asterisk to take in 5 hunting lines. Where one phone number would get published as our companies main IVR entry point, and the calls will get distributed into the Asterisk system internal extensions via the 5 available hunting lines. I am lost here. When a customer dials the main number, does it (A) get call transferred to an available channel by a dial plan with asterisk, or (B) the telco automatically checks to see if the main number is busy and transfer to the next hunting line? If (A), do I flash, dial to the available hunting line with my dial plan, and disconnect the original call (similar to a 3 way call conference). Would this even work on a external telco line? If (B), this would be simple, I would assume Asterisk can listen to all 5 fxo and run the same IVR script Here is my setup. 5 FXO hunting lines/19 FXS analog phone goes to a channel bank, then to a Digium T1 card. Thanks in advance, Sam ___ 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] Question about MP3's
Title: Message Hello all. I know * doesn't directly support recording mp3 files, but I was wondering if anyone has created an AGI to do it indirectly. Thanks in advance. B. J.
Re: [Asterisk-Users] Question about MP3's
On Mon, 2004-01-05 at 10:36, B. J. Bomar wrote: Hello all. I know * doesn't directly support recording mp3 files, but I was wondering if anyone has created an AGI to do it indirectly. Thanks in advance. That should be fairly trivial depending on what you want to accomplish. If all you want is audio in mp3 format in non realtime mode, just record to ulaw then send to sox or lame or some other app. If you want realtime, you might be able to rig an EAGI app to pipe audio to a app that compresses for you. Please reevaluate your need for MP3 as you will find that it doesn't compress as well as gsm when comparing file size with quality. I have done quite a bit of testing in this field as we record medical transcription where file size directly impacts performance and therefore profit. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] question re voicemail
Hi,
I just setup my * with digium. I started testing
voicemail first between atas, and i am not sure why it is not prompting me any
when the call is not answered or if busy.i only get continuous
ringback andthe following message:
asterisk*CLI --
Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new
stack -- Called 5104112978 --
SIP/5104112978-3f88 is ringing -- Nobody picked up in
2 ms
I wonder if my uextension and
bextension config is correct, mispelled, or something else is
missing. Note that ata to ata via * works,
as well as getting to VoicemailMain via extension 1234. Please
help. My config are found below. I appreciate your
help.
sip.conf
---
[6882332]type=friendusername=6882332secret=testhost=dynamicdefaultip=172.30.200.27dtmfmode=rfc2833mailbox=6882332callerid
= "test1" 6882332context=sip
[5104112978]type=friendusername=5104112978secret=testhost=dynamic;canreinvite=nodefaultip=172.30.200.26dtmfmode=rfc2833mailbox=5104112978callerid
= "test2" 5104112978context=sip
extensions.conf
[sip];ringexten =
5104112978,1,Dial(SIP/5104112978,20,tr)exten =
6882332,1,Dial(SIP/6882332,15,tr)exten =
,1,Dial(SIP/,5,tr)
;unansweredexten =
6882332,102,Voicemail,u6882332exten =
5104112978,102,Voicemail,u5104112978exten =
,102,Voicemail,u
;busyexten =
6882332,103,Voicemail,b6882332exten =
5104112978,103,Voicemail,b5104112978exten =
,103,Voicemail,b
;get messageexten =
1234,1,VoicemailMain(6882332);exten =
,1,VoicemailMain();
voicemail.conf
-
[default]6882332 =
6882332,test1,[EMAIL PROTECTED]
5104112978 = 5104112978,test2, [EMAIL PROTECTED]
9011 = 9011,Asterisk,[EMAIL PROTECTED]
= ,Nada,[EMAIL PROTECTED]
Re: [Asterisk-Users] question re voicemail
extensions.conf
[sip]
;ring
exten = 5104112978,1,Dial(SIP/5104112978,20,tr)
exten = 6882332,1,Dial(SIP/6882332,15,tr)
exten = ,1,Dial(SIP/,5,tr)
;unanswered
exten = 6882332,102,Voicemail,u6882332
exten = 5104112978,102,Voicemail,u5104112978
exten = ,102,Voicemail,u
;busy
exten = 6882332,103,Voicemail,b6882332
exten = 5104112978,103,Voicemail,b5104112978
exten = ,103,Voicemail,b
;get message
exten = 1234,1,VoicemailMain(6882332);
exten = ,1,VoicemailMain();
voicemail.conf
-
[default]
6882332 = 6882332,test1,[EMAIL PROTECTED]
5104112978 = 5104112978,test2, [EMAIL PROTECTED]
9011 = 9011,Asterisk,[EMAIL PROTECTED]
= ,Nada,[EMAIL PROTECTED]
I would do this:
exten = ,1,Dial(SIP/,5,tr)
exten = ,2,Voicemail,u
exten = ,102,Voicemail,b
Your priority numbering was off a little and its best to group the whole
extension together instead of spreading them out. Helps to make sure you
don't get lost. Also that 5 is in seconds.. not rings.
I would do this
[default]
exten = ,1,Macro(stdexten|)
exten = 6882332,1,Macro(stdexten|6882332)
exten = 5104112978,1,Macro(stdexten|5104112978)
[macro-stdexten]
exten = s,1,Dial(SIP/${ARG1},20,tr)
exten = s,2,Voicemail(u${ARG1})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG1})
exten = s,103,Hangup
Hope that helps.
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] question re voicemail
Hi Jess, It looks like your problem is with the extension increment. If there is no answer in the allotted time, the count increses by one. If the line is busy, the count increases by 101. Also, have you actually created the vm boxes you're referencing? Thanks! Sean -Original Message- From: Jess Magnaye [mailto:[EMAIL PROTECTED] Sent: Mon 1/5/2004 4:28 PM To: [EMAIL PROTECTED] Cc: Subject: [Asterisk-Users] question re voicemail Hi, I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message: asterisk*CLI -- Executing Dial(SIP/6882332-1697, SIP/5104112978|20|tr) in new stack -- Called 5104112978 -- SIP/5104112978-3f88 is ringing -- Nobody picked up in 2 ms I wonder if my uextension and bextension config is correct, mispelled, or something else is missing. Note that ata to ata via * works, as well as getting to VoicemailMain via extension 1234.Please help. My config are found below. I appreciate your help. sip.conf --- [6882332] type=friend username=6882332 secret=test host=dynamic defaultip=172.30.200.27 dtmfmode=rfc2833 mailbox=6882332 callerid = test1 6882332 context=sip [5104112978] type=friend username=5104112978 secret=test host=dynamic ;canreinvite=no defaultip=172.30.200.26 dtmfmode=rfc2833 mailbox=5104112978 callerid = test2 5104112978 context=sip extensions.conf [sip] ;ring exten = 5104112978,1,Dial(SIP/5104112978,20,tr) exten = 6882332,1,Dial(SIP/6882332,15,tr) exten = ,1,Dial(SIP/,5,tr) ;unanswered exten = 6882332,102,Voicemail,u6882332 exten = 5104112978,102,Voicemail,u5104112978 exten = ,102,Voicemail,u ;busy exten = 6882332,103,Voicemail,b6882332 exten = 5104112978,103,Voicemail,b5104112978 exten = ,103,Voicemail,b ;get message exten = 1234,1,VoicemailMain(6882332); exten = ,1,VoicemailMain(); voicemail.conf - [default] 6882332 = 6882332,test1,[EMAIL PROTECTED] 5104112978 = 5104112978,test2, [EMAIL PROTECTED] 9011 = 9011,Asterisk,[EMAIL PROTECTED] = ,Nada,[EMAIL PROTECTED] winmail.dat
[Asterisk-Users] QUESTION Ringing Appl.
Hello, I have a problem. When Idial to asterisk with H323 I do not hear ringing applecation (phone rings but i do not hear ringing tone in handset). I have tried with Cisco 2600 H323 and Quintum H323. But when I connect I can hear ringing appl. What can be wrong? Configuration is wrong? Please help! bart
Re: [Asterisk-Users] Question about incoming/outgoing
Larry Black wrote: [hardware] type=friend callerid=Hardware Phone 5 secret=phone echocancel=yes host=dynamic dtmfmode=rfc2833 context=sip My standard config for GS phones on the same LAN as the Asterisk server is.. [hardware] type = friend callerid = Hardware Phone 5 secret = phone host = dynamic dtmfmode = info context = sip To specify codecs you could add.. disallow = all allow = ulaw allow = alaw I can't see from you phone config why you are having this problem.. I know GS are close to a major upgrade on the firmware so maybe this will help.. Also a suggestion, you may find it easier to manage if you name the phones by their extension number rather than a name.. it will make your extensions.conf a little easier to create and modify.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question on hearing ADSI CAS tone
Hello all, Have a few questions. New to asterisk , just getting setup with 1 X100P and 2 TDM400p. Redhat 9 Hope I sent this to correct list Setting up some Aastra/Vista Powertouch 350 phones. Things work outofbox on ADSI programming, vmail downloads, menus etc. Question. When I Dial voicemail2 mainmenu from keypad on the 350 (Say ext 8) I hear what seems to be the CAS signal from the ADSI transmission. I hear the same thing at the end of the mailbox prompt and end of password prompt. Is it normal to be hearing this sound or should the phone be intercepting it? Prompts and everything works (as much as is programmed) but the sound is annoying as heck. I have noticed I hear this any time the phone is in voice mode, and is trying to switch into data mode for the next set of menu transmissions. I have no alternate equipment or lab or service to comparisons with. First experience with ADSI. So I am not sure if this is normal or not. BTW I also cannot get VMWI working as stated in current bug list. (VMWI broken on TDM400P) Thanks __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about incoming/outgoing
We've got one of the Budgetone phones here, and we can call from any SIP phone, or an outside line TO this phone and the conversation sounds great for bothways, not a bad delay, no echo problem, etc. But when we pick up the Budgetone and dial an outside line or another SIP phone the person on the Budgeton just sounds really choppy and there is a slight delay. We've messed with settings and tried each codec individually all with the same results. There is no problem with the network that the phone is on, its private set up to test this out. I'm not blameing the hardware, but I know i'm missing something. Calls to/from software SIP phones and a POTS phone sound great to/from SIP phone and outside lines. This is the only thing we're still having problems with. Has anyone run into this before or can anyone point me to any additional settings i can tweak ether on the phone or in asterisk? Thanks much. WB ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about incoming/outgoing
On Tue, Nov 18, 2003 at 05:29:25PM -0600, Wayne Black wrote: We've got one of the Budgetone phones here, and we can call from any SIP phone, or an outside line TO this phone and the conversation sounds great for bothways, not a bad delay, no echo problem, etc. But when we pick up the Budgetone and dial an outside line or another SIP phone the person on the Budgeton just sounds really choppy and there is a slight delay. We've messed with settings and tried each codec individually all with the same results. There is no problem with the network that the phone is on, its private set up to test this out. I'm not blameing the hardware, but I know i'm missing something. Calls to/from software SIP phones and a POTS phone sound great to/from SIP phone and outside lines. This is the only thing we're still having problems with. Has anyone run into this before or can anyone point me to any additional settings i can tweak ether on the phone or in asterisk? Thanks much. WB Can you send an example of your sip.conf for the GS phone to the list? If you have debugging on you might look through the debug log and see how the call is set up. Sounds like a codec problem. -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Question about incoming/outgoing
[hardware] type=friend callerid=Hardware Phone 5 secret=phone echocancel=yes host=dynamic dtmfmode=rfc2833 context=sip Larry D. Black CEO Black Sheep Computing, inc 2312 E Matthews Jonesboro, AR 72401 870.910.6969 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walker Haddock Sent: Tuesday, November 18, 2003 6:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Question about incoming/outgoing On Tue, Nov 18, 2003 at 05:29:25PM -0600, Wayne Black wrote: We've got one of the Budgetone phones here, and we can call from any SIP phone, or an outside line TO this phone and the conversation sounds great for bothways, not a bad delay, no echo problem, etc. But when we pick up the Budgetone and dial an outside line or another SIP phone the person on the Budgeton just sounds really choppy and there is a slight delay. We've messed with settings and tried each codec individually all with the same results. There is no problem with the network that the phone is on, its private set up to test this out. I'm not blameing the hardware, but I know i'm missing something. Calls to/from software SIP phones and a POTS phone sound great to/from SIP phone and outside lines. This is the only thing we're still having problems with. Has anyone run into this before or can anyone point me to any additional settings i can tweak ether on the phone or in asterisk? Thanks much. WB Can you send an example of your sip.conf for the GS phone to the list? If you have debugging on you might look through the debug log and see how the call is set up. Sounds like a codec problem. -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about IAX/DID's...
Hi, Here is a general question, not applying to asterisk so much, but in the application of asterisk. I have purchased a few IAX DID's through VoicePulse and am interested in a service provider who has the ability to provide me with one number (reliable, as I wish to publish), and the capacity to redirect those calls to my IAX DID's (is this even possible)? Also, with IAX DID's, how many calls per DID can an Asterisk box recieve? Is a DID, the same as one line? Or, can multiple people call into each DID at the same time? Regards, Phillip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question about codecs and interoperability with cisco AS5350
Hi all. I'm going to implement some large Asterisk based solution. Maybe 4-5 PCs with 1-2 E1/T1 trunks on each. Because some of the traffic will be sended to external VoIP provider, i has some questions 1. Which is the lowest bandwidth consuming codec in Asterisk, which is compatible with Cisco Gateways. Stability is needed too. 2. Have someone allready bulded such a systems and what hardware (pc i mean) is needed Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
