[Asterisk-Users] modules.conf again
Hi everybody and sorry for posting this again to the list. I don't want you guys to think that I'm DEMANDING FOR SUPPORT (we all have just had several discusions about that) but my experience tell me that if a posted question is easy enough, it is answered immediatly, or it will never be answerd!!!(people forget it...) Mine are two very simple questions about modules.conf. It's only a problem of concept, perhaps a wrong idea, only that. I please ask you to answer, because for me it's important to have everything more or less clear. THANKS VERY MUCH in advance, and here they are, my two little questions... 1)As I have seen, to make Asterisk load chan_capi.so and chan_modem.so you must have: load=>chan_capi.so and load => chan_modem.so in your modules.conf. But I had understood some time ago that setting autoload => yes made Asterisk load every module that was necesary. Then, why must I load these channels explicitely? 2)For what is used the section [global]? thanks a lot one more time cmayor ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error on PRI channel : Call specified but not found!
Anyone encountered this error on an PRI channel local PTSN: WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 18 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 19 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 19 Cristian VASILIU mail: [EMAIL PROTECTED] www : cvasiliu.home.ro Soon sip address. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ignorepat doesn't work
Hi
in order to keep the dial tone after pressing 9 for 'outside line' I
have this in my extensions.conf
[localpstn]
ignorepat => 9
exten => _9[123456789]XXX,1,Dial,${PSTN}/${EXTEN:1}
exten => _9[123456789]XXX,2,Congestion
this is properly included in the handsets' context but the dial tone
disappears after pressing 9.
am I missing something?
thanks in advance
regards
bk
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Re: [Asterisk-Users] Transfert call
Yeah ! It is good I will try now Rattana - Original Message - From: "carlos del mayor" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, July 08, 2003 5:22 PM Subject: Re: [Asterisk-Users] Transfert call Sorry! Didn't know it got implemented!!Last notice I had is that it would be implemented soon, but didn't think it was SO soon...Great then!! cmayor --- Martin Pycko <[EMAIL PROTECTED]> escribió: > That got implemented recently ... > > Martin > > On Tue, 8 Jul 2003, carlos del mayor wrote: > > > Hi Rattana, > > > > That kind of transfer is not yet implemented in *. > The > > way it will be indicated is: > > exten =>111,dial,Zap/1,20,T > > > > The T indicate that transfer is permitted for > calling > > party, but as I've said, that's not implemented at > the > > moment. > > > > Regards > > cmayor > > > > --- Rattana BIV <[EMAIL PROTECTED]> escribió: > > Hi, > > > > > > > > > A question about transfert. > > > > > > How can I make transfert with the the person who > > > call. > > > X call Z and X transfert Z to Y. > > > I only succeed to do X call Z and Z transfert to > Y. > > > > > > If someone have a solution it will be very good > =) > > > > > > > > > regards > > > Rattana > > > > > ___ > > Yahoo! Messenger - Nueva versión GRATIS > > Super Webcam, voz, caritas animadas, y más... > > http://messenger.yahoo.es > > ___ > > 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 ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ 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] Error on PRI channel : Call specified but not found!
Anyone encountered this error on an PRI channel local PTSN: WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 18 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 19 WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not found? WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad channel 19 In the zapata.conf configuration file I have the following line : immediate=no! I have to specify the with immediate=yes it is working even if it is asking for the s into the pri context! Cristian VASILIU mail: [EMAIL PROTECTED] www : cvasiliu.home.ro Soon sip address. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbe Questions.
Dear all, I'm just finished installing the TDM (2 port) and X100P. I'm using X100P to pstn, and the TDM to the phone. I've loaded the module, and I can also list the card in the /proc/zaptel/ I'm a little confused now. in zapatel.conf, how do I know which channel is which. (TDM or X100P)? Thanks and pardon my English Isianto Istiadi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] modules.conf again
Il mer, 2003-07-09 alle 09:31, carlos del mayor ha scritto: > THANKS VERY MUCH in advance, and here they are, my two > little questions... Well, here are my two little answers, I hope they are not too wildly incorrect :) > 1)As I have seen, to make Asterisk load chan_capi.so > and chan_modem.so you must have: load=>chan_capi.so > and load => chan_modem.so in your modules.conf. But I > had understood some time ago that setting autoload => > yes made Asterisk load every module that was necesary. > Then, why must I load these channels explicitely? Because that way they are loaded first. Some other modules use symbols that are exported by those modules, and if those other modules got loaded first, they wouldn't work. > 2)For what is used the section [global]? My sample cfg says: ; Module names listed in "global" section will have symbols globally ; exported to modules loaded after them. So it is needed to get the aforementioned result! Bye, -- E. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_capi hanging channels
Hi, Using chan_capi 0.2.2 and Asterisk CVS-06/16/03-15:55:17 with a AVM Fritz card.. For some reason both the channels (show channels) were in a Down state.. I had to retart * to clear the channels and recieve calls again.. This is the second time it has happened.. Anyone got any ideas how to fix it?? or what I should check?? Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error on web page for msn
Hi everybody, I'm trying to use msn with * and for that, I'm reading all information on the mailing list. You used to recommend the page http://mcleod.pbx.nq.net/msn/, but I always get an error while opening. Has it changed? Is there another one? Thanks cmayor ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] modules.conf again
thanks a lot, E., now I understand it! regards cmayor --- Emanuele Pucciarelli <[EMAIL PROTECTED]> escribió: > Il mer, 2003-07-09 alle 09:31, carlos del mayor ha > scritto: > > THANKS VERY MUCH in advance, and here they are, my > two > > little questions... > > Well, here are my two little answers, I hope they > are not too wildly > incorrect :) > > > 1)As I have seen, to make Asterisk load > chan_capi.so > > and chan_modem.so you must have: > load=>chan_capi.so > > and load => chan_modem.so in your modules.conf. > But I > > had understood some time ago that setting autoload > => > > yes made Asterisk load every module that was > necesary. > > Then, why must I load these channels explicitely? > > Because that way they are loaded first. Some other > modules use symbols > that are exported by those modules, and if those > other modules got > loaded first, they wouldn't work. > > > 2)For what is used the section [global]? > > My sample cfg says: > > ; Module names listed in "global" section will have > symbols globally > ; exported to modules loaded after them. > > So it is needed to get the aforementioned result! > > Bye, > > -- > E. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Matching winth asterisk-oh323
Hi, It is possible to do matching in oh323.conf with asterisk-oh323? example : alias=0XXX Regards Rattana
Re: [Asterisk-Users] Chan_capi hanging channels
WipeOut . wrote: Hi, Using chan_capi 0.2.2 and Asterisk CVS-06/16/03-15:55:17 with a AVM Fritz card.. For some reason both the channels (show channels) were in a Down state.. I had to retart * to clear the channels and recieve calls again.. This is the second time it has happened.. Anyone got any ideas how to fix it?? or what I should check?? Thanks.. I had same problem - just rolled back to 0.2.1a -- - Best Regards, Pavel Litvinenko. ICQ: 16224754 Ph: (8632) 923962, 923640 sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterix Manual
Hi, I am new to Asterix. I would like to know from where I can get a manual on how to use Asterix. Thanks Dhammika
Re: [Asterisk-Users] voip
On Monday 07 July 2003 10:40 pm, [EMAIL PROTECTED] wrote: > On Tue, 8 Jul 2003, marrandy wrote: > > > Well I now have asterisk installed. > > > > I've printed out the asterisk web site. > > > > I've printed the draft Asterisk handbook V2 > > > > I've printed the Introduction to the asterisk open source pbx > > oh, no, the trees, the humanity... try and not print, its easier to cut > paste from a console than print the buggers out, besides the pace of * > development at times has been so fast, those printed pages are probably > obsolete by now :) > I'm so busy I pretty much have to print 'dead trees' on major piecies of software. I end up reading them as I travel and have time, make notes etc. Just the way it is. Some of us don't have the luxury of sitting around all day with spare time on their hands to study new topics and applications. Get over it. -- If you're not very clever you should be conciliatory. -- Benjamin Disraeli ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterix Manual
http://www.digium.com/handbook-draft.pdf Dhammika Gunawardena (ISP) wrote: Hi, I am new to Asterix. I would like to know from where I can get a manual on how to use Asterix. Thanks Dhammika ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Use dialing plan from h.323 gatekeeper?
Hi, I want to configure * to use a gatekeeper for routing calls to H.323 endpoints. I imagine it will work like that: * (chan_h323) will query the gatekeeper where to terminate the dialed number and the gatekeeper will return the information for the h.323 gateway. after that chan_h323 will try to make the call to the gateway it has received from the gatekeeper. so instead of duplicating a gatekeeper's dialing plan (with all registered gateways and terminals) in *, can I just configure * to receive the endpoint for terminating a call from the gatekeeper for lets say all calls starting with prefix 9? Hristo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] It's true - Nikotel charge for not-completed calls
Hi A few days ago, Kelly remarked that he had previously observed that Nikotel charged him for calls he did not actually complete. I have made a number of test calls to my landline without picking up the calls. I just let it ring once and hung up on the calling phone. A look at the call records on MyNikotel reveals that I was charged six seconds for every of these calls. I have raised a support request with Nikotel to ask for this to be corrected. However, Kelly said he had done so before and never got any reply. Therefore, I would like to ask everybody on the list who is using Nikotel to verify this for themselves and raise a support request on Nikotel's website. Together we may be able to get this fixed. thanks regards benjamin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Matching winth asterisk-oh323
Rattana BIV wrote: Hi, It is possible to do matching in oh323.conf with asterisk-oh323? example : alias=0XXX No. In this case you will put in oh323.conf: prefix=0 and then, in extensions, you will do the pattern matching you want. Regards Rattana Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 prob :)
Dave Alan Caruana wrote:
i'm getting Asterisk to dial an h323 call termination service ..
right now getting this message:
-- Executing Wait("Zap/1-1", "1") in new stack
-- Accepting call from '21382890' to 's' on channel 1, span 1
-- Executing Dial("Zap/1-1", "OH323/h323:[EMAIL PROTECTED]") in new
stack
5:59.330 H323 Cleaner H323Connection
ip$localhost/18729 terminated.
ERROR[1230546240]: File chan_oh323.c, Line 704 (oh323_call): H323:0: Could
not call h323:[EMAIL PROTECTED]
-- Couldn't call h323:[EMAIL PROTECTED]
-- Hungup 'H323:0'
== Everyone is busy at this time
-- Executing Hangup("Zap/1-1", "") in new stack
== Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
any idea what that can mean ?
In oh323.conf set
libTraceLevel=3
and rerun (WARNING: you will get tons of messages)
Send the output (preferably directly to me) to check what
happened.
I have my system currently working through SIP, however every now and then
it shows this
message
-- Got SIP response 481 "Invalid CSeq Number" back from 216.52.153.207
== Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1'
and drops the line which is the reason I am trying to use H323 instead,
maybe I can
get around that problem. Can anyone tell me what it means?
thanks
Dave
Michael.
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Re: [Asterisk-Users] error on web page for msn
ok, sorry about that folks http://ausfone.com/msn/ should now be used instead ;-) Gary On Wed, 9 Jul 2003 11:29:17 +0200 (CEST), carlos del mayor wrote: >Hi everybody, >I'm trying to use msn with * and for that, I'm reading >all information on the mailing list. You used to >recommend the page http://mcleod.pbx.nq.net/msn/, but >I always get an error while opening. Has it changed? >Is there another one? >Thanks >cmayor > >___ >Yahoo! Messenger - Nueva versi¢n GRATIS >Super Webcam, voz, caritas animadas, y m s... >http://messenger.yahoo.es >___ >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
RE: [Asterisk-Users] Use dialing plan from h.323 gatekeeper?
Hi, I think I understood how to achieve this. Anyway, a working config is welcome if anyone has already done it. hristo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of HT Sent: Wednesday, July 09, 2003 2:54 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Use dialing plan from h.323 gatekeeper? Hi, I want to configure * to use a gatekeeper for routing calls to H.323 endpoints. I imagine it will work like that: * (chan_h323) will query the gatekeeper where to terminate the dialed number and the gatekeeper will return the information for the h.323 gateway. after that chan_h323 will try to make the call to the gateway it has received from the gatekeeper. so instead of duplicating a gatekeeper's dialing plan (with all registered gateways and terminals) in *, can I just configure * to receive the endpoint for terminating a call from the gatekeeper for lets say all calls starting with prefix 9? Hristo ___ 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] PBX / Asterisk integration
Hi all, I regret that I don't know much about telephony as I'm a networking bod, but here goes... We are thinking about implementing a VoIP service so that staff and students can make VoIP calls from home or using our wireless LAN on campus. Clearly, we would like it to integrate with our PBX so VoIP users can talk to the PSTN as well. We don't actually control the University's PBX, so it is highly desirable that any changes to the PBX are very minimal! We were thinking of setting up a single extension which is associated with a huntgroup on the PBX that somehow connects (via an E1?) to the Asterisk box. If a user leaves his office he sets his extension to re-direct calls to that extension number (or to re-direct on no-reply). The PBX would route the call to the asterisk huntgroup. Asterisk would "see" the original extension number called (assuming this is possible!!), and (knowing which extension maps to which user to which IP address) route the call to the user over the IP network. I can handle the IP stuff without any problems. My question is: is this a possible/sensible approach to implementing this type of service? If not, what's a better solution? TIA for any suggestions/comments, josh. -- --- Josh Howlett, Networking & Digital Communications, Information Systems & Computing, University of Bristol, U.K. 'phone: 0117 928 7850 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat doesn't work
Hi bk,
On Wed, Jul 09, 2003 at 17:16:55 +0900, BK [address only for mailing lists] wrote:
> Hi
>
> in order to keep the dial tone after pressing 9 for 'outside line' I
> have this in my extensions.conf
>
> [localpstn]
> ignorepat => 9
> exten => _9[123456789]XXX,1,Dial,${PSTN}/${EXTEN:1}
> exten => _9[123456789]XXX,2,Congestion
>
> this is properly included in the handsets' context but the dial tone
> disappears after pressing 9.
>
> am I missing something?
I had the same problem here and discovered that "ignorepat" only works if
it's placed in the actual incoming context of your channels and not if
it's included from another context. Not sure if this is a bug or a feature.
So, try placing the "ignorepat" in your handset-contexts instead.
Grtz,
Oliver
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Re: [Asterisk-Users] error on web page for msn
woops... http://phone.nq.net/msn might actually find it... it wil be moving soon to http://www.ausfone.com/msn after some reorganisation here... On Wed, 9 Jul 2003 11:29:17 +0200 (CEST), carlos del mayor wrote: >Hi everybody, >I'm trying to use msn with * and for that, I'm reading >all information on the mailing list. You used to >recommend the page http://mcleod.pbx.nq.net/msn/, but >I always get an error while opening. Has it changed? >Is there another one? >Thanks >cmayor > >___ >Yahoo! Messenger - Nueva versi¢n GRATIS >Super Webcam, voz, caritas animadas, y m s... >http://messenger.yahoo.es >___ >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
RE: [Asterisk-Users] Net2Phone SIP
Just to say that I've now managed to get this going by pretending to be an ATA 186. change the User-Agent string in chan_sip.c from "Asterisk" to "Cisco ATA 186" and the Net2Phone Sip service works with * Would it be possible to pick this up from sip.conf in a future release? Regards Mark -Original Message- From: Mark Thompson Sent: 02 June 2003 21:11 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Net2Phone SIP I've been trying to use net2phone's sip service at sip.net2phone.com with * but keep getting SIP/2.0 401 Unauthorized. Do you know if this should be possible? So far: I can use an ata186 to connected directly to n2p through sip.net2phone.com without any special settings. I can connect from * to iconnecthere, but, whatever I try from * to n2p produces "SIP/2.0 401 Unauthorized" (Can forward the full * sip log and ata186 log if it would help) Many thanks Mark -- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.55:5060;branch=z9hG4bK7d701e63 From: "112" ;tag=as16e11aa8 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 149 v=0 o=root 9698 9698 IN IP4 10.0.0.55 s=session c=IN IP4 10.0.0.55 t=0 0 m=audio 19622 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 (no NAT) to 66.33.146.12:5060 -- Called [EMAIL PROTECTED] Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.55:5060;branch=z9hG4bK7d701e63 From: "112" ;tag=as16e11aa8 To: ;tag=3ed78add-12013 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: "net2phone" User-Agent: Asterisk PBX Content-Length: 0 ___ 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
Re: [Asterisk-Users] modules.conf again
> 1)As I have seen, to make Asterisk load chan_capi.so > and chan_modem.so you must have: load=>chan_capi.so > and load => chan_modem.so in your modules.conf. But I > had understood some time ago that setting autoload => > yes made Asterisk load every module that was necesary. > Then, why must I load these channels explicitely? I can't speak for chan_capi, but in the case of chan_modem, it has to be explicitly listed because explicitly listed modules come first, and chan_modem will choose which modules it wants to load (it's some ugly historical baggage). > 2)For what is used the section [global]? Unfortunately, I don't know of any way to, in the module ".so" file itself, tell the linker to make its symbols global when it loads it. This causes Asterisk to specifically request the symbols be made global (and thus available to other modules which are loaded later). Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error on PRI channel : Call specified but notfound!
That's a new one, but contact Martin and he should be able to help assist you. Mark On Wed, 9 Jul 2003, Cristi wrote: > Anyone encountered this error on an PRI channel local PTSN: > WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad > channel 18 > WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, > but not found? > WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad > channel 19 > WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, > but not found? > WARNING[9226]: File chan_zap.c, Line 5830 (pri_dchannel): Hangup on bad > channel 19 > In the zapata.conf configuration file I have the following line : > immediate=no! I have to specify the with immediate=yes it is working > even if it is asking for the s into the pri context! > > > Cristian VASILIU > mail: [EMAIL PROTECTED] > www : cvasiliu.home.ro > Soon sip address. > > ___ > 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
Re: [Asterisk-Users] Chan_capi hanging channels
WipeOut . napsal(a): Hi, Using chan_capi 0.2.2 and Asterisk CVS-06/16/03-15:55:17 with a AVM Fritz card.. For some reason both the channels (show channels) were in a Down state.. I had to retart * to clear the channels and recieve calls again.. This is the second time it has happened.. Anyone got any ideas how to fix it?? or what I should check?? Thanks.. Sometimes I have same problems with Zap channels. Petr Michalek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterix Manual
Here you can find some documentation: http://www.digium.com/index.php?menu=documentation Regards cmayor --- "Dhammika Gunawardena (ISP)" <[EMAIL PROTECTED]> escribió: > Hi, > I am new to Asterix. > I would like to know from where I can get a manual > on how to use Asterix. > > Thanks > Dhammika > ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PBX / Asterisk integration
> I regret that I don't know much about telephony as I'm a networking bod, > but here goes... > > We are thinking about implementing a VoIP service so that staff and > students can make VoIP calls from home or using our wireless LAN on > campus. > > Clearly, we would like it to integrate with our PBX so VoIP users can > talk to the PSTN as well. > > We don't actually control the University's PBX, so it is highly > desirable that any changes to the PBX are very minimal! Hi, we're currently trying to do something like that here at Saarland University. We will give each VoIP-User his own extension that will be routed to Astersik over an E1 (We will use extensions like 69XXX for that). I think this routing can be done easily in the PBX. We will also set up a SIP Express Router and give an account to every student here. Students will eventually also get their own extension that can be reached from everywhere (something like 70X). The students will be able to call every phone here on campus over the Asterisk-System (including other students of course :) ). This is mainly to prove that a simple cheap VoIP system can handle that traffic. In a second step we will try to replace the existing PBX with Asterisk and SIP phones or channel banks. This will of course take some time. :) Another goal of the project is to route calls to other universities over IP if they support it. You can find some information about our project on our Website (see below). -- Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP pgp0.pgp Description: PGP signature
[Asterisk-Users] Music on hold quality..
Hi, Does any one have any pointers on improving moh quality?? Symptoms are crackling and hissing as the sound comes and goes.. I installed mpg123 this morning.. I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they all sounded terrible... The PC is a P4 so its got plenty of processing power.. I have tried a few different types of classical music (Piano, Violin and full Orchestra).. Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] more abou msn
Hi, Talking about messenger,,, it's still necesary to do HKEY_CURRENT_USER\Software\Microsoft\MessengerService\Corp2PC_Phone equals to '1' ??? But it's still sending the '+' digit, so it's necesary to stripMSD? Thanks a lot cmayor ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] caller id
Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is [EMAIL PROTECTED] I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on hold quality..
On Wed, 2003-07-09 at 08:57, WipeOut . wrote: > Hi, > > Does any one have any pointers on improving moh quality?? > > Symptoms are crackling and hissing as the sound comes and goes.. > > I installed mpg123 this morning.. > > I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they > all sounded terrible... The PC is a P4 so its got plenty of processing > power.. I have tried a few different types of classical music (Piano, > Violin and full Orchestra).. Is this on Zap channels, or VoIP. IF VoIP do you have some form of Zap channel on your system? -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ignorepat doesn't work
thanks On Wednesday, July 9, 2003, at 10:07 PM, The Traveller wrote: I had the same problem here and discovered that "ignorepat" only works if it's placed in the actual incoming context of your channels and not if it's included from another context. thinking about it, this makes sense because there may be multiple contexts with extensions starting with the same ignorepat digit. Not sure if this is a bug or a feature. probably intentional. So, try placing the "ignorepat" in your handset-contexts instead. Well, it works now on the Zap channels but not on the SIP phones. Does anyone know how to fix this for SIP phones? but it's not that important anyway. thanks again rgds bk
[Asterisk-Users] callerid= being ignored
Hi
I have defined my SIP phones like this ...
[Sip1]
username=gs1
callerid= "Full name" <1001>
etc etc
Now, when I do this in a given extension
exten => ,1,NoOp(${CALLERIDNUM})
then I get "" as callerid and not "<1001>" as defined with callerid=
Sure, I could set the usernames to their respective extensions, but I
don't want to do that. I'd like to keep login names independent of
extensions.
Is there any fix for this so that the real callerid shows up?
thanks in advance
regards
bk
___
Asterisk-Users mailing list
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Re: [Asterisk-Users] Music on hold quality..
No I don't have any Zap channels, I am using chan_capi for my PSTN connection and SIP only.. Does moh depend on zap?? > On Wed, 2003-07-09 at 08:57, WipeOut . wrote: > > Hi, > > > > Does any one have any pointers on improving moh quality?? > > > > Symptoms are crackling and hissing as the sound comes and goes.. > > > > I installed mpg123 this morning.. > > > > I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they > > all sounded terrible... The PC is a P4 so its got plenty of processing > > power.. I have tried a few different types of classical music (Piano, > > Violin and full Orchestra).. > > > Is this on Zap channels, or VoIP. IF VoIP do you have some form of Zap > channel on your system? > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip jitter buffer
This is kind of a repost of one part of a previous question I have had. Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter Format 213.137.73.178 xx 3705df0a5f7 00103/0 0ms ms 4 1 active SIP channel(s) I see that there is 0ms Jitter set. How can I set a Jitter buffer for use with sip channels? I can't seem to find any documentation about this. Any help is always appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP-H323v2 transcoder?
Hello, I have some MGCP VoIP gateways and some H323v2 VoIP gateways, Can a use the Asterisk for interconnect the VoIP boxes? If I can anyone knows how to configure it? Thank you very much Best regards Sebastian Sill. Uruguay. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming callerid on FXO
Hi my Digium FXO card isn't picking up the callerid I get from the PSTN. I have verified with a deskphone that can display the callerid that the facility works. So, it's definitely the FXO card not picking it up. As I am in Japan, I guess that NTT uses a different method to provide the callerid and so I guess that it is just a matter of configuring the FXO card so that it uses the right method for detection. I seem to remember that I read somewhere that this can be changed but I can't seem to find any reference to that now. Does anybody know how to change the method for detecting callerid? thanks in advance regards bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323, Asterisk and DTMF issue
Hi folks, Im using chan_h323 to dial out to a gateway which connects me to the PSTN. In order to use a menu system such my bank menu system, I have to set dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info wont work with Asterisks voicemail system. Im using the g.729 codec for h323 and Asterisk. Im told dtmfmode=inband wont work with g.729. Is it possible to use dtmfmode=info with h323 and access my Asterisk voicemail? Summary: dtmfmode = info ; works with h323 not with Asterisk Voicemail dtmfmode = inband ; works with h323 (with a flood of warnings) not with Asterisk Voicemail dtmfmode = rfc2833 ; works with Asterisk Voicemail not with h323 Any help would be greatly appreciated. Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Accurate Billing
My friend has a small pay phone services business running using analog lines and a conventional PBX. He allows a delay before starting billing. So if a customer's call is not answered after the allowed delay he is billed. Shepherd _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on hold quality..
On Wed, 2003-07-09 at 10:12, WipeOut . wrote: > No I don't have any Zap channels, I am using chan_capi for my PSTN connection and > SIP only.. > > Does moh depend on zap?? Yes, it uses Zap for timing. Look into ztdummy, or the rtc driver. > > On Wed, 2003-07-09 at 08:57, WipeOut . wrote: > > > Hi, > > > > > > Does any one have any pointers on improving moh quality?? > > > > > > Symptoms are crackling and hissing as the sound comes and goes.. > > > > > > I installed mpg123 this morning.. > > > > > > I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they > > > all sounded terrible... The PC is a P4 so its got plenty of processing > > > power.. I have tried a few different types of classical music (Piano, > > > Violin and full Orchestra).. > > > > > > Is this on Zap channels, or VoIP. IF VoIP do you have some form of Zap > > channel on your system? > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music on hold quality..
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: 09 July 2003 16:13 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold quality.. No I don't have any Zap channels, I am using chan_capi for my PSTN connection and SIP only.. Does moh depend on zap?? > On Wed, 2003-07-09 at 08:57, WipeOut . wrote: > > Hi, > > > > Does any one have any pointers on improving moh quality?? > > > > Symptoms are crackling and hissing as the sound comes and goes.. > > > Installing zaprtc from http://www.junghanns.net/asterisk/ will correct the problem. MOH uses the zaptel devices for timings - without any in the system you will experience poor quality playback. -nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] incoming callerid on FXO
Have not used this but in zaptel/wcfxo.c there is a constant JAPAN that if set claims to handle NTT -Original Message- From: BK [address only for mailing lists] <[EMAIL PROTECTED]> To: Asterisk List <[EMAIL PROTECTED]> Date: July 9, 2003 8:46 AM Subject: [Asterisk-Users] incoming callerid on FXO >Hi > >my Digium FXO card isn't picking up the callerid I get from the PSTN. > >I have verified with a deskphone that can display the callerid that the >facility works. So, it's definitely the FXO card not picking it up. > >As I am in Japan, I guess that NTT uses a different method to provide >the callerid and so I guess that it is just a matter of configuring the >FXO card so that it uses the right method for detection. I seem to >remember that I read somewhere that this can be changed but I can't seem >to find any reference to that now. > >Does anybody know how to change the method for detecting callerid? > >thanks in advance >regards >bk > >___ >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
Re: [Asterisk-Users] Music on hold quality..
> On Wed, 2003-07-09 at 10:12, WipeOut . wrote: >> No I don't have any Zap channels, I am using chan_capi for my PSTN >> connection and SIP only.. >> >> Does moh depend on zap?? > > Yes, it uses Zap for timing. Look into ztdummy, or the rtc driver. > Is there any documentation on using the ztdummy or rtc driver? Other than the note to remove the # from the Makefile I haven't been able to find any documentation on how to get the drivers working. - Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SUMMARY: Problems with Hangup Detection in VoiceMail2.
Many thanks to Martin Pycko and Mark Spencer. Mark's suggestion below was correct: "Maybe it's stuck trying to send the e-mail notification. If you take the e-mail address out of /etc/asterisk/voicemail.conf does that speed it up?" Indeed it did! The problem turned out to be a 60second delay while invoking mail, caused by a mis-configuration of my hostname and /etc/mail/sendmail.cf (must have been waiting while some DNS lookups timed out). My original problem is below: >Hi. >Has anyone experienced hangup detection problems with the VoiceMail2 app? >I have a console phone on the FXS port. When I call a SIP phone, and get >its voicemail greeting, I can enter the VoiceMail2 app, leave a message, >and then hit # to stop message recording. > >Recording does stop, but the channel stays up inside the VoiceMail2 app >(as shown by a "show channels" command) for about 60 seconds. Then it >drops. > >Is this expected behavior? If not, how can I make it disconnect the >call sooner? > >I get the very same behavior in all scenarios: > a) if calling from a console phone to a SIP phone. > b) if calling in on a POTS line, connected to an FXO card. > >a portion of my extensions.conf file is shown below: > . > . >exten => 304,1,Dial(SIP/304,16) ;fred >exten => 304,2,Voicemail2(u304) >exten => 304,3,Hangup > . > . > -- Regards, Fred -_=o&o>_--- Fred R. Ziegler BMWOA #77929 AMA #631103 <[EMAIL PROTECTED]> BMWRA #22468 '97 R1100RTL W. Medford, MA 02155YB #235 IBA #6190 '72 R60/5 swb ... holder of the Ultra-Prestigious MA IBMWR plate!! :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] caller id
Use SetCallerID(1234567). Tan telappliant.com - Original Message - From: "Marian Danisek" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 09, 2003 3:23 PM Subject: [Asterisk-Users] caller id Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is [EMAIL PROTECTED] I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ 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
Re: [Asterisk-Users] MGCP-H323v2 transcoder?
Sure RTFM Jeremy McNamara Sebastian Sill wrote: Hello, I have some MGCP VoIP gateways and some H323v2 VoIP gateways, Can a use the Asterisk for interconnect the VoIP boxes? If I can anyone knows how to configure it? Thank you very much Best regards Sebastian Sill. Uruguay. ___ 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
Re: [Asterisk-Users] callerid= being ignored
At the moment asterisk can get the callerid from the "From: " field.
regards
Martin
On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote:
>
> Hi
>
> I have defined my SIP phones like this ...
>
> [Sip1]
> username=gs1
> callerid= "Full name" <1001>
>
> etc etc
>
> Now, when I do this in a given extension
>
> exten => ,1,NoOp(${CALLERIDNUM})
>
> then I get "" as callerid and not "<1001>" as defined with callerid=
>
> Sure, I could set the usernames to their respective extensions, but I
> don't want to do that. I'd like to keep login names independent of
> extensions.
>
> Is there any fix for this so that the real callerid shows up?
>
> thanks in advance
> regards
> bk
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
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Re: [Asterisk-Users] Music on hold quality..
On Wed, 2003-07-09 at 11:13, Joe Cooke wrote: > > On Wed, 2003-07-09 at 10:12, WipeOut . wrote: > >> No I don't have any Zap channels, I am using chan_capi for my PSTN > >> connection and SIP only.. > >> > >> Does moh depend on zap?? > > > > Yes, it uses Zap for timing. Look into ztdummy, or the rtc driver. > > > > Is there any documentation on using the ztdummy or rtc driver? Other than > the note to remove the # from the Makefile I haven't been able to find any > documentation on how to get the drivers working. Basically you load the driver, the driver produces the appropriate interrupts needed to keep MOH happy. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E1-RJ45 pin configuration
hi! We have ISDN/PRI E1 lines which needs to be connected to the E400P card . Can somebody help us with the PIN configuration of RJ45 in relation with the E1(ISDN/PRI) ? urs, DenZel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] It's true - Nikotel charge for not-completed calls
Unless iconnecthere has changed something recently, they do the same thing. I dialed a number without the leading "1" and ich tried to make an international call. No one answered because it was an invalid number but they still charged me for an international call. - Original Message - From: "BK [address only for mailing lists]" <[EMAIL PROTECTED]> To: "Asterisk List" <[EMAIL PROTECTED]> Sent: Wednesday, July 09, 2003 8:08 AM Subject: [Asterisk-Users] It's true - Nikotel charge for not-completed calls > Hi > > A few days ago, Kelly remarked that he had previously observed that > Nikotel charged him for calls he did not actually complete. > > I have made a number of test calls to my landline without picking up the > calls. I just let it ring once and hung up on the calling phone. > > A look at the call records on MyNikotel reveals that I was charged six > seconds for every of these calls. > > I have raised a support request with Nikotel to ask for this to be > corrected. However, Kelly said he had done so before and never got any > reply. > > Therefore, I would like to ask everybody on the list who is using > Nikotel to verify this for themselves and raise a support request on > Nikotel's website. Together we may be able to get this fixed. > > thanks > regards > benjamin > > ___ > 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
Re: [Asterisk-Users] E1-RJ45 pin configuration
On Wed, 2003-07-09 at 14:22, denzel fernando wrote: > hi! > > We have ISDN/PRI E1 lines which needs to be connected to the E400P card > . Can somebody help us with the PIN configuration of RJ45 in relation > with the E1(ISDN/PRI) ? Have you searched the archive? Have you used google? The E400P uses standard wiring. If you are coming from your telco to the E400P, you should be able to use standard cat5 wired the same way you do for ethernet. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI with variable length numbers
Hey all, I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming into it from a Meridian-switch. The incoming numbers on this PRI all start with the same digit and the last part of the dialled number is signalled to Asterisk digit by digit, until Asterisk signals that the number is complete and the call rings. All works well, unless I have 2 or more numbers which start with the same digits. In that case, dialling will be signalled as complete as soon as the shortest of the numbers is dialed. An example: exten => 31801,1, ... exten => 3180,1, ... In this case, dialling "3180" will immediately start ringing, while on similar setups, with TDM40B's and analog phones, Asterisk will wait for the duration of "digittimeout" for more digits, if it can't be absolutely certain that the dialled number is complete. If I remove the "3180" in the above example, ringing will only start after "31801" is fully dialled. Relevant configs: /etc/zaptel.conf: bchan=1-15 dchan=16 bchan=17-31 /etc/asterisk/zapata.conf: switchtype=euroisdn signalling=pri_cpe immediate=no Any ideas? Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] It's true - Nikotel charge for not-completed calls
Guys, unless their site states something to the contrary you don't have a hope in hell with this. you are paying them for a voip circuit, which you are using to attempt a call. you have taken up several seconds of voip bandwidth which they are charging you for, the same way you would pay if there is 6 seconds of silence during your phone conversation. (This takes roughly the same amount of transfer as an unsuccessful call setup.) By the same reasoning, if you have some strange audio in your conversation which does not compress well, they don't charge you extra since it costs more to deliver your call than you are paying. The price is based on averages. Do you complain at an all you can eat buffet if you get full after a small plateful ? Same concept. At 01:39 PM 7/9/2003 -0400, you wrote: Unless iconnecthere has changed something recently, they do the same thing. I dialed a number without the leading "1" and ich tried to make an international call. No one answered because it was an invalid number but they still charged me for an international call. - Original Message - From: "BK [address only for mailing lists]" <[EMAIL PROTECTED]> To: "Asterisk List" <[EMAIL PROTECTED]> Sent: Wednesday, July 09, 2003 8:08 AM Subject: [Asterisk-Users] It's true - Nikotel charge for not-completed calls > Hi > > A few days ago, Kelly remarked that he had previously observed that > Nikotel charged him for calls he did not actually complete. > > I have made a number of test calls to my landline without picking up the > calls. I just let it ring once and hung up on the calling phone. > > A look at the call records on MyNikotel reveals that I was charged six > seconds for every of these calls. > > I have raised a support request with Nikotel to ask for this to be > corrected. However, Kelly said he had done so before and never got any > reply. > > Therefore, I would like to ask everybody on the list who is using > Nikotel to verify this for themselves and raise a support request on > Nikotel's website. Together we may be able to get this fixed. > > thanks > regards > benjamin > > ___ > 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI with variable length numbers
Why do you see the problem with that ? How would you use that functionality that analog channels have ? With PRI you're going to always receive the full number that was dialed. So since you have a limited number of DID's do the matching one for one and it'll work. regards Martin On Wed, 9 Jul 2003, The Traveller wrote: > Hey all, > > I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming > into it from a Meridian-switch. The incoming numbers on this PRI all start > with the same digit and the last part of the dialled number is signalled to > Asterisk digit by digit, until Asterisk signals that the number is > complete and the call rings. > > All works well, unless I have 2 or more numbers which start with the same > digits. In that case, dialling will be signalled as complete as soon > as the shortest of the numbers is dialed. An example: > > exten => 31801,1, ... > exten => 3180,1, ... > > In this case, dialling "3180" will immediately start ringing, while > on similar setups, with TDM40B's and analog phones, Asterisk will > wait for the duration of "digittimeout" for more digits, if it can't > be absolutely certain that the dialled number is complete. If I remove > the "3180" in the above example, ringing will only start after "31801" > is fully dialled. > > > Relevant configs: > > /etc/zaptel.conf: > > bchan=1-15 > dchan=16 > bchan=17-31 > > /etc/asterisk/zapata.conf: > > switchtype=euroisdn > signalling=pri_cpe > immediate=no > > > Any ideas? > > > > Grtz, > > Oliver > ___ > 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
Re: [Asterisk-Users] callerid= being ignored
On Thursday, July 10, 2003, at 01:31 AM, Martin Pycko wrote:
At the moment asterisk can get the callerid from the "From: " field.
Thanks,
I tried fromuser=1001 in the [Sip1] section and CALLERIDNUM still
returns ""
What is the "callerid=" directive good for if not setting the CALLERID,
it doesn't seem to make any sense.
Does anybody know how to fix this?
thanks in advance
regards
bk
On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote:
I have defined my SIP phones like this ...
[Sip1]
username=gs1
callerid= "Full name" <1001>
etc etc
Now, when I do this in a given extension
exten => ,1,NoOp(${CALLERIDNUM})
then I get "" as callerid and not "<1001>" as defined with
callerid=
Sure, I could set the usernames to their respective extensions, but I
don't want to do that. I'd like to keep login names independent of
extensions.
Is there any fix for this so that the real callerid shows up?
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[Asterisk-Users] Asterisk basic how-to on O'Reilly's site
This was published on O'Reilly's site last week, but they didn't tell me until now. :) The article is pretty minimal, because I had a limited number of words to work with, so many features are not implemented. However, it's a good start. http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html I had a few people review it beforehand, but I'd also welcome changes or comments on it, as your comments will help me in future articles that I may write on the subject. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as SIP <-> PSTN gateway
Hi, I'm new to Asterisk and have a couple of basic questions. We're interested in using * simply as a SIP <-> PSTN gateway using a T400P connected to one or more ISDN PRI lines (instead of using a Cisco box which would cost more and come with no hackable source code :-) First, is Asterisk's SIP stack up to date and fully functional with respect to the SIP protocol? Are there any known limitations that would cause problems in this application? Is there a general 'to-do' list for SIP support? E.g., it doesn't seem that chan_sip.c supports SUBSCRIBE/NOTIFY DTMF events, which some Cisco boxes seem generate for DTMF (?). Secondly, roughly what kind of CPU/system horsepower would required to support transferring 96 channels of voice data between SIP/Ethernet and the PRI if: (a) if no transcoding were being performed (i.e., both the RTP pkts and PRI B-channels were carrying ulaw data); and (b) transcoding from e.g. ulaw on the PRI <-> GSM in the RTP. Thanks, -Archie __ Archie Cobbs *Halloo Communications* http://www.halloo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] experience with multi-port SIP/FXS gateways?
I'm proposing an asterisk configuration and considering the use of multiport SIP/FXS gateways (instead of T1 cards and channel banks). I'm looking for products similar in function to the Cisco ATA-186, but with more ports. I've seen the manufacturer's web pages for the Audiocodes MediaPack (http://www.audiocodes.com/) and the Mediatrix (http://www.mediatrix.com/) access devices. Does anyone have any experiences to share with these or similar devices, or opinions on the relative merits of gateways vs channel banks? Thanks very much! John [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] It's true - Nikotel charge for not-completed calls
On Thursday, July 10, 2003, at 02:59 AM, Jon Pounder wrote: Guys, unless their site states something to the contrary you don't have a hope in hell with this. you are paying them for a voip circuit, which you are using to attempt a call. you have taken up several seconds of voip bandwidth which they are charging you for, the same way you would pay if there is 6 seconds of silence during your phone conversation. (This takes roughly the same amount of transfer as an unsuccessful call setup.) By the same reasoning, if you have some strange audio in your conversation which does not compress well, they don't charge you extra since it costs more to deliver your call than you are paying. The price is based on averages. Do you complain at an all you can eat buffet if you get full after a small plateful ? Same concept. First of all, nobody complained. As far as your argument is concerned, it can be dismissed very easily because of a) common sense; and b) about 120 years of telephony history during which it has been customary to charge for payload not for effort VoIP is not going to change this well established business practise If Nikotel gets enough requests, they are likely to rectify their billing, if not, sooner or later some competitor is trying to get a competitive edge by making it their sales pitch that they don't charge for unconnected time. Just like by-six-second billing or even by-the-second billing has eventually won over by-30-seconds or by-the-minute billing. It is that simple. Your reasoning notwithstanding. rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI with variable length numbers
Hey Martin, I'm not receiving fixed-length numbers on that PRI and it really seems to be the Asterisk end which decides when dialling is complete. I've arranged for a block of numbers, starting with "3", to be routed from the Meridian to Asterisk, over this PRI. As long as the numbers I set up in my dialplan don't start with the same digits, I can put in numbers with varying lengths and it works OK, dialling them from an extension on the Meridian. Don't ask me how, because I'm not really into the low-level signalling-protocols on PRI's yet. I could send you "pri intense debug"-output of some calls off-list, if you want to see for yourself. For example, I have set up several extensions in the range of "3000" to "3099" for IP-phones, and those can all be dialled directly. I also included a context in the incoming PRI-context which allowed dialling IAXTel-numbers, with a prefix of "3", and you can now indeed call "31700" from any extension on the Meridian for an IAXTel-number. The Meridian has no special programming to differentiate between dialled extensions in the "3"-range, so my conclusion is that Asterisk signals when dialling is complete. As Asterisk seems to be the one determining when the number is finished, it seems possible to use the same extension matching-logic as on analog ports. Grtz, Oliver On Wed, Jul 09, 2003 at 13:25:33 -0500, Martin Pycko wrote: > Why do you see the problem with that ? > > How would you use that functionality that analog channels have ? > > With PRI you're going to always receive the full number that was dialed. > So since you have a limited number of DID's do the matching one for one > and it'll work. > > regards > Martin > > On Wed, 9 Jul 2003, The Traveller wrote: > > > Hey all, > > > > I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming > > into it from a Meridian-switch. The incoming numbers on this PRI all start > > with the same digit and the last part of the dialled number is signalled to > > Asterisk digit by digit, until Asterisk signals that the number is > > complete and the call rings. > > > > All works well, unless I have 2 or more numbers which start with the same > > digits. In that case, dialling will be signalled as complete as soon > > as the shortest of the numbers is dialed. An example: > > > > exten => 31801,1, ... > > exten => 3180,1, ... > > > > In this case, dialling "3180" will immediately start ringing, while > > on similar setups, with TDM40B's and analog phones, Asterisk will > > wait for the duration of "digittimeout" for more digits, if it can't > > be absolutely certain that the dialled number is complete. If I remove > > the "3180" in the above example, ringing will only start after "31801" > > is fully dialled. > > > > > > Relevant configs: > > > > /etc/zaptel.conf: > > > > bchan=1-15 > > dchan=16 > > bchan=17-31 > > > > /etc/asterisk/zapata.conf: > > > > switchtype=euroisdn > > signalling=pri_cpe > > immediate=no > > > > > > Any ideas? > > > > > > > > Grtz, > > > > Oliver > > ___ > > 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323 v0.5.3
Hi Michael, are you adding ilbc support to your channel ? On Tuesday 08 Jul 2003 12:07 pm, Michael Manousos wrote: > Hello all, > > I have updated the asterisk-oh323 package. The new version > has several improvements (fixes in audio/RTP stream generation, > music-on-hold working, flash hook detection, more config options). > You can download it from: > > http://www.inaccessnetworks.com/projects/asterisk-oh323 > > Feedback is always welcome. > > Regards, > Michael. > > > ___ > 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
Re: [Asterisk-Users] incoming callerid on FXO
Hi, Mine pick up my own phone number as callerid. What can be done? I am located in Romania. PSTN system is based on Siemens switchboard. Dan - Original Message - From: "BK [address only for mailing lists]" <[EMAIL PROTECTED]> To: "Asterisk List" <[EMAIL PROTECTED]> Sent: Wednesday, July 09, 2003 6:21 PM Subject: [Asterisk-Users] incoming callerid on FXO > Hi > > my Digium FXO card isn't picking up the callerid I get from the PSTN. > > I have verified with a deskphone that can display the callerid that the > facility works. So, it's definitely the FXO card not picking it up. > > As I am in Japan, I guess that NTT uses a different method to provide > the callerid and so I guess that it is just a matter of configuring the > FXO card so that it uses the right method for detection. I seem to > remember that I read somewhere that this can be changed but I can't seem > to find any reference to that now. > > Does anybody know how to change the method for detecting callerid? > > thanks in advance > regards > bk > > ___ > 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] OpenBSD version???
John Todd's Onlamp article mentions an OpenBSD version as of June 2003. Have I been sleeping while reading asterisk-users? Is it a seperate project or is it just making Asterisk portable? Who is working on this and is it in the main CVS yet? Do they have device drivers ported or just the software parts of Asterisk? I think the software parts would be relatively simple but time consuming. I've been trying to work up the nerve to try a port to FreeBSD, but I don't have a lot of time and haven't been a C coder for many years now. Anything the OpenBSD people have done will probably make a FreeBSD port trivial. I have nothing against OpenBSD, and while Linux is acceptable, I have FreeBSD boxen laying around all over the place doing other tasks with more than enough spare ooomph to handle Asterisk. I could roll out 4 VoIP installations tomorrow, with PSTN tie-ins to follow, if it would run on FreeBSD. -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Call Manager doc
I am looking for a doc out there that on how to use the Asterisk Call Manager. Can someone let me know what the URL to this is. Thanks, John Haigh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H450 problems
Hello people, I am using Asterisk with a handful of Micronet SP5100 IP Phones and a Micronet SP5052 FXO Gateway. So far I have incoming calls ringing all the phones correctly, outgoing calls working, voicemail working and calls between phones working. The only think I cant get working is Transferring of calls, and the ability to put calls on hold. The phones have both a 'Transfer' button and a 'Hold' button on them. I have tried using both the H323 and oh323 modules, but neither seems to work. My extensions are setup like so: exten => 8000,1,Dial(h323/10.0.3.11,20,T)exten => 8000,2,Answer exten => 8000,3,Playback(call-later) exten => 8000,4,Hangup I have also tried using hook-flash to do transfers, but it just hangs up on the caller. I have read as much documentation as I can find, and have been lurking in IRC for days!
Re: [Asterisk-Users] incoming callerid on FXO
I have this problem before ... make sure you DON'T have callerid set in zapata.conf else all incoming callerid are set to zapata.conf's $callerid. Sunny. On Wed, 2003-07-09 at 12:59, Dan wrote: > Hi, > > Mine pick up my own phone number as callerid. > What can be done? > I am located in Romania. > PSTN system is based on Siemens switchboard. > > Dan > > - Original Message - > From: "BK [address only for mailing lists]" <[EMAIL PROTECTED]> > To: "Asterisk List" <[EMAIL PROTECTED]> > Sent: Wednesday, July 09, 2003 6:21 PM > Subject: [Asterisk-Users] incoming callerid on FXO > > > > Hi > > > > my Digium FXO card isn't picking up the callerid I get from the PSTN. > > > > I have verified with a deskphone that can display the callerid that the > > facility works. So, it's definitely the FXO card not picking it up. > > > > As I am in Japan, I guess that NTT uses a different method to provide > > the callerid and so I guess that it is just a matter of configuring the > > FXO card so that it uses the right method for detection. I seem to > > remember that I read somewhere that this can be changed but I can't seem > > to find any reference to that now. > > > > Does anybody know how to change the method for detecting callerid? > > > > thanks in advance > > regards > > bk > > > > ___ > > 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 -- Sunny Woo <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to modify dialed number?
Hi! Is there simple way how to add prefix to dialed number? I need change 0X. to 0X. Regards Petr Michálek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Warning
When starting *, I get the following when the chan_iax2.so loads: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found WARNING[16384]: File chan_iax2.c, Line 4980 (set_config): Ignoring port for now == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 As I am not loading chan_iax.so and IAX is working, it is obviously not fatal, but after a reasonable amount of checking docs, archives etc. I am curious to know why it occurs and it is not using port 5036 as specified in iax.conf. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to modify dialed number?
exten => _0X,1,Dial,Zap/g1/0${EXTEN:1}
Martin
On Wed, 9 Jul 2003, Petr Michálek wrote:
> Hi!
>
> Is there simple way how to add prefix to dialed number?
> I need change 0X. to 0X.
>
> Regards
>
> Petr Michálek
>
> ___
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> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [Asterisk-Users] How to modify dialed number?
On Wednesday 09 July 2003 04:37 pm, Petr Michálek wrote: > Is there simple way how to add prefix to dialed number? > I need change 0X. to 0X. See the applications StripMSD and Prefix. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone and Voicemail
I have a Budgetone 102 with the latest firmware 1.0.3.72 and using dtmfmode=rfc2833 With g711 I have no problem with Voicemail or Voicemail2. With g729 it always repeats digits and it is impossible to check my voicemail (or any other apps that require digits) - Original Message - From: "WipeOut ." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, July 08, 2003 3:42 PM Subject: Re: [Asterisk-Users] Budgetone and Voicemail > I have had double digits being passed every now and then once I am into voicemail.. I haven't had a problem with the initial login stage.. I also haven't had time to look into it yet.. > > You could try changing the DTMF mode and see if it helps.. > > Later.. > > > I have a problem with using voicemail on the Budgetone phones. When > > entering the mailbox and password, sometimes some keys will register > > multiple times (as shown on console when it says no such user in config > > file) and sometimes some keys won't even register at all. It seems > > totally random. Has anyone seen this problem? Any recommendations > > would be greatly appreciated. Thanks. > > > > > > Brian Borders > > [EMAIL PROTECTED] > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > __ > http://www.linuxmail.org/ > Now with e-mail forwarding for only US$5.95/yr > > Powered by Outblaze > ___ > 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
Re: [Asterisk-Users] Budgetone and Voicemail
try dtmfmode=info solved all my former problems and even works with g723.1 :) On Wednesday 09 Jul 2003 11:15 pm, Dan Fernandez wrote: > I have a Budgetone 102 with the latest firmware 1.0.3.72 and using > dtmfmode=rfc2833 > > With g711 I have no problem with Voicemail or Voicemail2. > > With g729 it always repeats digits and it is impossible to check my > voicemail (or any other apps that require digits) > > > - Original Message - > From: "WipeOut ." <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, July 08, 2003 3:42 PM > Subject: Re: [Asterisk-Users] Budgetone and Voicemail > > > I have had double digits being passed every now and then once I am into > > voicemail.. I haven't had a problem with the initial login stage.. I also > haven't had time to look into it yet.. > > > You could try changing the DTMF mode and see if it helps.. > > > > Later.. > > > > > I have a problem with using voicemail on the Budgetone phones. When > > > entering the mailbox and password, sometimes some keys will register > > > multiple times (as shown on console when it says no such user in config > > > file) and sometimes some keys won't even register at all. It seems > > > totally random. Has anyone seen this problem? Any recommendations > > > would be greatly appreciated. Thanks. > > > > > > > > > Brian Borders > > > [EMAIL PROTECTED] > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > __ > > http://www.linuxmail.org/ > > Now with e-mail forwarding for only US$5.95/yr > > > > Powered by Outblaze > > ___ > > 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to modify dialed number?
On Thursday, July 10, 2003, at 06:37 AM, Petr Michálek wrote:
Is there simple way how to add prefix to dialed number?
I need change 0X. to 0X.
how about this
exten => 0X.,1,Dial(0{EXTEN:1})
rgds
bk
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[Asterisk-Users] IAX G729 Codec
Hi, I have recently purchased some Asterisk G729 Codecs and installed them to overcome by bandwidth problem I was having with GSM. The G729 keeps the pings nice and low, but the audio stutters or jitters a fair bit. (Starts and stops) Any Idea what would be causing this ? I am just testing it using the OSS/Console at the moment, as I am waiting for my Digium cards to arrive. Thanks Jay
Re: [Asterisk-Users] How to modify dialed number?
You forgot about "_" in front of 0X
Martin
On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote:
>
> On Thursday, July 10, 2003, at 06:37 AM, Petr Michálek wrote:
>
> > Is there simple way how to add prefix to dialed number?
> > I need change 0X. to 0X.
>
> how about this
>
> exten => 0X.,1,Dial(0{EXTEN:1})
>
> rgds
> bk
>
>
> ___
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> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [Asterisk-Users] IAX2 Warning
IAX2 uses hardcoded 4569 port so it's not looking for port keyword. Nothing to worry about. Martin On Thu, 10 Jul 2003, Richard Scobie wrote: > When starting *, I get the following when the chan_iax2.so loads: > > [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) > == Manager registered action IAXpeers > == Parsing '/etc/asterisk/iax.conf': Found > WARNING[16384]: File chan_iax2.c, Line 4980 (set_config): Ignoring port > for now > == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) > == Using TOS bits 16 > == IAX Ready and Listening on 0.0.0.0 port 4569 > > As I am not loading chan_iax.so and IAX is working, it is obviously not > fatal, but after a reasonable amount of checking docs, archives etc. I > am curious to know why it occurs and it is not using port 5036 as > specified in iax.conf. > > Regards, > > Richard > > > ___ > 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
Re: [Asterisk-Users] Budgetone and Voicemail
Yes! It did work with g729 and dtmfmode=info. Thanks a lot! - Original Message - From: "Michael Bielicki" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 09, 2003 7:35 PM Subject: Re: [Asterisk-Users] Budgetone and Voicemail > try dtmfmode=info > solved all my former problems and even works with g723.1 > :) > On Wednesday 09 Jul 2003 11:15 pm, Dan Fernandez wrote: > > I have a Budgetone 102 with the latest firmware 1.0.3.72 and using > > dtmfmode=rfc2833 > > > > With g711 I have no problem with Voicemail or Voicemail2. > > > > With g729 it always repeats digits and it is impossible to check my > > voicemail (or any other apps that require digits) > > > > > > - Original Message - > > From: "WipeOut ." <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, July 08, 2003 3:42 PM > > Subject: Re: [Asterisk-Users] Budgetone and Voicemail > > > > > I have had double digits being passed every now and then once I am into > > > > voicemail.. I haven't had a problem with the initial login stage.. I also > > haven't had time to look into it yet.. > > > > > You could try changing the DTMF mode and see if it helps.. > > > > > > Later.. > > > > > > > I have a problem with using voicemail on the Budgetone phones. When > > > > entering the mailbox and password, sometimes some keys will register > > > > multiple times (as shown on console when it says no such user in config > > > > file) and sometimes some keys won't even register at all. It seems > > > > totally random. Has anyone seen this problem? Any recommendations > > > > would be greatly appreciated. Thanks. > > > > > > > > > > > > Brian Borders > > > > [EMAIL PROTECTED] > > > > > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > > __ > > > http://www.linuxmail.org/ > > > Now with e-mail forwarding for only US$5.95/yr > > > > > > Powered by Outblaze > > > ___ > > > 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 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
Re: [Asterisk-Users] IAX G729 Codec
On Wed, 2003-07-09 at 18:06, Jay Tyndall wrote: > Hi, > > I have recently purchased some Asterisk G729 Codecs and installed them > to overcome by bandwidth problem I was having with GSM. > > The G729 keeps the pings nice and low, but the audio stutters or > jitters a fair bit. (Starts and stops) > > Any Idea what would be causing this ? I am just testing it using the > OSS/Console at the moment, as I am waiting for my Digium cards to > arrive. No zapata device for timing would probably be one thing. Also, what kind of network are you crossing? I have found some problems when the network is not instantaneous, but not internet level lagging. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H450 problems
Hello Aaron Martin, Yesterday we discussed the possibility to buy some Micronet hardware. We haven't used hardware from this vendor yet and I wanted to ask you a question about the Micronet SP5052 FXO Gateway features. Is there any way to setup SP5052 FXO to answer a call from PSTN and to forward it immediately to the Asterisk box - using hotline dial or something like that. PSTN -> Micronet SP5052 FXO -> Asterisk IVR (Thank you for calling etc) Best regards Lubo Aaron Martin wrote: Hello people, I am using Asterisk with a handful of Micronet SP5100 IP Phones and a Micronet SP5052 FXO Gateway. So far I have incoming calls ringing all the phones correctly, outgoing calls working, voicemail working and calls between phones working. The only think I cant get working is Transferring of calls, and the ability to put calls on hold. The phones have both a 'Transfer' button and a 'Hold' button on them. I have tried using both the H323 and oh323 modules, but neither seems to work. My extensions are setup like so: exten => 8000,1,Dial(h323/10.0.3.11,20,T) exten => 8000,2,Answer exten => 8000,3,Playback(call-later) exten => 8000,4,Hangup I have also tried using hook-flash to do transfers, but it just hangs up on the caller. I have read as much documentation as I can find, and have been lurking in IRC for days! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H450 problems
Yes this is exactly what I am doing. A caller dials in on the PSTN, and gets connected to the Asterisk server, which answers with "Please dial the extension you require". The caller then dials 8000 and gets transferred to extension 8000. However, I cant seem to get transfer working, ie if they call 8000 but were actually after 8001, I want the user on extension 8000 to be able to tranfer the call to 8001.. Please help someone!! - Original Message - From: "Lubomir Christov" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 10, 2003 2:19 PM Subject: Re: [Asterisk-Users] H450 problems > Hello Aaron Martin, > > Yesterday we discussed the possibility to buy some Micronet hardware. We > haven't used hardware from this vendor yet and I wanted to ask you a > question about the Micronet SP5052 FXO Gateway features. > Is there any way to setup SP5052 FXO to answer a call from PSTN and to > forward it immediately to the Asterisk box - using hotline dial or > something like that. > > PSTN -> Micronet SP5052 FXO -> Asterisk IVR (Thank you for calling etc) > > Best regards > Lubo > > > > Aaron Martin wrote: > > Hello people, > > > > I am using Asterisk with a handful of Micronet SP5100 IP Phones and a > > Micronet SP5052 FXO Gateway. > > > > So far I have incoming calls ringing all the phones correctly, outgoing > > calls working, voicemail working and calls between phones working. The > > only think I cant get working is Transferring of calls, and the ability > > to put calls on hold. > > > > The phones have both a 'Transfer' button and a 'Hold' button on them. > > > > I have tried using both the H323 and oh323 modules, but neither seems to > > work. My extensions are setup like so: > > > > exten => 8000,1,Dial(h323/10.0.3.11,20,T) > > exten => 8000,2,Answer > > exten => 8000,3,Playback(call-later) > > exten => 8000,4,Hangup > > > > I have also tried using hook-flash to do transfers, but it just hangs up > > on the caller. > > > > I have read as much documentation as I can find, and have been lurking > > in IRC for days! > > ___ > 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
RE: [Asterisk-Users] Asterix Manual
thanks -Original Message- From: carlos del mayor [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 09, 2003 7:50 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterix Manual Here you can find some documentation: http://www.digium.com/index.php?menu=documentation Regards cmayor --- "Dhammika Gunawardena (ISP)" <[EMAIL PROTECTED]> escribió: > Hi, > I am new to Asterix. > I would like to know from where I can get a manual > on how to use Asterix. > > Thanks > Dhammika > ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ 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
Re: [Asterisk-Users] Billsec on CDR
Steve Can you please give us some guidance on how to make call progress work outside the US or UK? Thanks Dan - Original Message - From: "Stephen Davies" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, June 21, 2003 4:51 AM Subject: Re: [Asterisk-Users] Billsec on CDR > > > On Fri, 20 Jun 2003, Tan Aks wrote: > > > Isn't there any way to make callprogress work for people in Europe? What is > > it that is needed to make it work? > > I've done call progress for the UK. Patch to the -dev list by the end > of the weekend. > > What country do you want? > > Steve > > > ___ > 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
Re: [Asterisk-Users] IAX G729 Codec
This is going across a 256k/64 to a 512k/128. They are about 2 hops away from each other and ping times are sub 70ms. (Even when the * audio is playing) - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 10, 2003 12:02 PM Subject: Re: [Asterisk-Users] IAX G729 Codec > On Wed, 2003-07-09 at 18:06, Jay Tyndall wrote: > > Hi, > > > > I have recently purchased some Asterisk G729 Codecs and installed them > > to overcome by bandwidth problem I was having with GSM. > > > > The G729 keeps the pings nice and low, but the audio stutters or > > jitters a fair bit. (Starts and stops) > > > > Any Idea what would be causing this ? I am just testing it using the > > OSS/Console at the moment, as I am waiting for my Digium cards to > > arrive. > > No zapata device for timing would probably be one thing. > > Also, what kind of network are you crossing? I have found some problems > when the network is not instantaneous, but not internet level lagging. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > 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
Re: [Asterisk-Users] How to modify dialed number?
On Thursday, July 10, 2003, at 08:13 AM, Martin Pycko wrote:
You forgot about "_" in front of 0X
Indeed, and I also forgot to put a channel (Zap, SIP, IAX ...) in the
dial string.
Oh dear, I guess I have to go to bed a bit earlier ;-)
rgds
bk
On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote:
On Thursday, July 10, 2003, at 06:37 AM, Petr Michálek wrote:
Is there simple way how to add prefix to dialed number?
I need change 0X. to 0X.
how about this
exten => 0X.,1,Dial(0{EXTEN:1})
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Re: [Asterisk-Users] IAX G729 Codec
On Wed, 2003-07-09 at 23:30, Jay Tyndall wrote: > This is going across a 256k/64 to a 512k/128. > They are about 2 hops away from each other and ping times are sub 70ms. > (Even when the * audio is playing) What kind of slow as hell routers are you using? The ping times from my office to home is in the 50ms range and at 17 hops and a tour of the south east US. According to mapquest, using just city names, my route is ~2k miles one way. BTW, my problems where on our private T1 line that sees round trips in the 4ms range. Our semi educated guess was that we had a problem with the jitter buffer causing echo cancel to go nutty when our ping times would occasionally jump to 20ms. When I turned off the jitter buffer, the call quality became so clear that people don't believe we are VoIP. > - Original Message - > From: "Steven Critchfield" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Thursday, July 10, 2003 12:02 PM > Subject: Re: [Asterisk-Users] IAX G729 Codec > > > > On Wed, 2003-07-09 at 18:06, Jay Tyndall wrote: > > > Hi, > > > > > > I have recently purchased some Asterisk G729 Codecs and installed them > > > to overcome by bandwidth problem I was having with GSM. > > > > > > The G729 keeps the pings nice and low, but the audio stutters or > > > jitters a fair bit. (Starts and stops) > > > > > > Any Idea what would be causing this ? I am just testing it using the > > > OSS/Console at the moment, as I am waiting for my Digium cards to > > > arrive. > > > > No zapata device for timing would probably be one thing. > > > > Also, what kind of network are you crossing? I have found some problems > > when the network is not instantaneous, but not internet level lagging. > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > 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 -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] incoming callerid on FXO
Hi Sunny, I don't have it. There is nowhere specified in the config files my own PSTN phone number, so it takes it from the PSTN line. Thanks, Dan - Original Message - From: "Sunny Woo" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 10, 2003 12:03 AM Subject: Re: [Asterisk-Users] incoming callerid on FXO > I have this problem before ... make sure you DON'T have callerid set in > zapata.conf else all incoming callerid are set to zapata.conf's > $callerid. > > Sunny. > > On Wed, 2003-07-09 at 12:59, Dan wrote: > > Hi, > > > > Mine pick up my own phone number as callerid. > > What can be done? > > I am located in Romania. > > PSTN system is based on Siemens switchboard. > > > > Dan > > > > - Original Message - > > From: "BK [address only for mailing lists]" <[EMAIL PROTECTED]> > > To: "Asterisk List" <[EMAIL PROTECTED]> > > Sent: Wednesday, July 09, 2003 6:21 PM > > Subject: [Asterisk-Users] incoming callerid on FXO > > > > > > > Hi > > > > > > my Digium FXO card isn't picking up the callerid I get from the PSTN. > > > > > > I have verified with a deskphone that can display the callerid that the > > > facility works. So, it's definitely the FXO card not picking it up. > > > > > > As I am in Japan, I guess that NTT uses a different method to provide > > > the callerid and so I guess that it is just a matter of configuring the > > > FXO card so that it uses the right method for detection. I seem to > > > remember that I read somewhere that this can be changed but I can't seem > > > to find any reference to that now. > > > > > > Does anybody know how to change the method for detecting callerid? > > > > > > thanks in advance > > > regards > > > bk > > > > > > ___ > > > 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 > -- > Sunny Woo <[EMAIL PROTECTED]> > > ___ > 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
