Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
Hi, On Tue, November 29, 2005 13:50 Francesco Peeters wrote: BTW: BRIstuff is not included by default as it breaks PRI support. Asterisk is already set up to use zap, so that is easy... As far as I know, BRIstuff is not included for licencing reasons... Is it true, that PRI support and BRIstuff are now incompatible? (In version 1.0.9 I had no problems to use two HFC cards and one TE110 in one system). Thank You for any informations on that topic. Best regards Karsten Wemheuer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WG: App_rxfax problem
Dunno :) what do you thing is wrong there? the compile was fine! I only need a solution how to fix this error!! On Sat, 03 Dec 2005 01:52:03 +0800 Steve Underwood [EMAIL PROTECTED] wrote: How could a CVS update fix an error you have made during installation? Steve René Enskat [Teamware GmbH] wrote: so is there a solution in the next cvs udpate? *Von:* René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] *Gesendet:* Donnerstag, 1. Dezember 2005 14:47 *An:* 'asterisk-users@lists.digium.com' *Betreff:* WG: App_rxfax problem I just sent the error in full log: Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so]Dec 1 15:01:08 WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so failed! *Von:* René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] *Gesendet:* Donnerstag, 1. Dezember 2005 08:35 *An:* 'asterisk-users@lists.digium.com' *Betreff:* App_rxfax problem When i load the fax modules into the asterisk i got this errors but compile was ok! I have the latest cvs head [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Sat, December 3, 2005 9:28, Karsten Wemheuer said: Hi, On Tue, November 29, 2005 13:50 Francesco Peeters wrote: BTW: BRIstuff is not included by default as it breaks PRI support. Asterisk is already set up to use zap, so that is easy... As far as I know, BRIstuff is not included for licencing reasons... Is it true, that PRI support and BRIstuff are now incompatible? (In version 1.0.9 I had no problems to use two HFC cards and one TE110 in one system). I have seen it stated on several sites that it is incompatible, but I have also heard from different people that it does work... I do not have a PRI, so I cannot comment on it. It is possible that they've included that because actually getting it to work may be a hit and miss thing (correct loading orders, etc.), but again, I cannot comment on that as I do not have a PRI! ;-) All I have are problems with the BRI... :-o (But that may (partially) be the Dutch KPN's fault as well, insisting on bringing down the D channel every minute... I never understood the reasoning behind it, as bringing down a running connection that often only increases chances of incompatability, like the 3Com NetBuilder had, and the chances of instability, like I have now...) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't compile chan_zap.c on fresh cvs 1.20 checkout
I tried compiling Asterisk 1.20 (fresh cvs checkout of just some 15 minutes ago) on a CentOS 4.2 box. Compiling zaptel seems to work fine but when I try to compile * I get this: chan_zap.c:8904: error: structure has no member named `useruserinfo' chan_zap.c:8012: warning: unused variable `plancallingani' chan_zap.c: In function `handle_pri_show_debug': chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug' chan_zap.c: In function `setup_zap': chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this function) chan_zap.c: In function `load_module': chan_zap.c:10870: warning: passing arg 1 of `pri_set_error' from incompatible pointer type chan_zap.c:10871: warning: passing arg 1 of `pri_set_message' from incompatible pointer type make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 Ideas anyone? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iax2 connection failed
HI:
i tried to send calls to callshopcompany
(www.callshopcompany.com) using iax2 but the call
fails giving me this:
dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
iax2/callshopcompany/0017046872001) in new stack
-- Called callshopcompany/0017046872001
-- Call accepted by 213.61.187.150 (format g729)
-- Format for call is g729
-- Hungup 'IAX2/callshopcompany/2'
== No one is available to answer at this time
Dec 3 11:09:21 WARNING[23053]: pbx.c:1949
ast_pbx_run: Timeout, but no rule 't' in context
'calls'
Hangup on console
i have this my extensions.conf:
[calls]
exten = _00.,1,Dial,iax2/callshopcompany/${EXTEN}
;
i have this my iax.conf :
[callshopcompany]
type=peer
host=213.61.187.150
username=X
secret=X
disallow=all
allow=g729
__
Start your day with Yahoo! - Make it your home page!
http://www.yahoo.com/r/hs
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voip providers with trunked iax2
Hi list: Can any body gives me a voip provider with trunked iax2 ,because i have tried voipjet and sixtel and they are not trunked . Regards; chawki __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Conf Realtime?
Hi I've been playing around with Realtime Asterisk using the ODBC module to connect to my database and I got extensions working but now I'm looking to get my iax.conf into the database. I would like to have the users who can register with my box to dial extensions in there, and also the connections to my outbound providers (voicepulse, voxee, etc etc). I've been trying to read the wiki but it doesnt really have a good documentation of what I need to do with my current iax.conf to tell it to look in the database for things. Do I have to use switch = Realtime like I do in extensions.conf? Any help or information here is appreciated. Thanks alot! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax2 connection failed
jonny hashem a écrit :
HI:
i tried to send calls to callshopcompany
(www.callshopcompany.com) using iax2 but the call
fails giving me this:
dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
iax2/callshopcompany/0017046872001) in new stack
-- Called callshopcompany/0017046872001
-- Call accepted by 213.61.187.150 (format g729)
-- Format for call is g729
-- Hungup 'IAX2/callshopcompany/2'
== No one is available to answer at this time
Dec 3 11:09:21 WARNING[23053]: pbx.c:1949
ast_pbx_run: Timeout, but no rule 't' in context
'calls'
The call doesn't fail: nobody pickup and call finish with timeout. See
below how to get away from the message
Hangup on console
i have this my extensions.conf:
[calls]
exten = _00.,1,Dial,iax2/callshopcompany/${EXTEN}
exten = t,1,Hangup
;
i have this my iax.conf :
[callshopcompany]
type=peer
host=213.61.187.150
username=X
secret=X
disallow=all
allow=g729
--
Daniel
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax2 connection failed
Hi:
Now the time out is message is gone ,why the call
still fails?
--- Administrator TOOTAI [EMAIL PROTECTED] wrote:
jonny hashem a écrit :
HI:
i tried to send calls to callshopcompany
(www.callshopcompany.com) using iax2 but the call
fails giving me this:
dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
iax2/callshopcompany/0017046872001) in new stack
-- Called callshopcompany/0017046872001
-- Call accepted by 213.61.187.150 (format
g729)
-- Format for call is g729
-- Hungup 'IAX2/callshopcompany/2'
== No one is available to answer at this time
Dec 3 11:09:21 WARNING[23053]: pbx.c:1949
ast_pbx_run: Timeout, but no rule 't' in context
'calls'
The call doesn't fail: nobody pickup and call
finish with timeout. See
below how to get away from the message
Hangup on console
i have this my extensions.conf:
[calls]
exten = _00.,1,Dial,iax2/callshopcompany/${EXTEN}
exten = t,1,Hangup
;
i have this my iax.conf :
[callshopcompany]
type=peer
host=213.61.187.150
username=X
secret=X
disallow=all
allow=g729
--
Daniel
___
--Bandwidth and Colocation provided by Easynews.com
--
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
__
Start your day with Yahoo! - Make it your home page!
http://www.yahoo.com/r/hs
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] prepaid application
Just a quick note to say thanks to all who replied, most helpful. Thanks Again Scott scott wrote: Hi All I am using prepaid auth (callingcards), the idea is for a prepaid support line. It is up and running but I have a couple of questions with regards to modifications I would like to make. When a user calls and they go through the process of entering their card number. They are then asked for a destination. What I would like to be able to do is not have it ask for a destination and automatically dial a number? At present I ask them to enter a default number when it ask for a destination and this then takes them to a queue, if someone is available it rings and goes through, if no one is available rather than sit in the queue and listen to the lovely onhold music prepaid auth comes back and says that destination is unreachable, is there a way to get it to just wait in the queue? Many Thanks In Advance Scott Pinhorne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax2 connection failed
chawki hammoud a écrit : Hi: Now the time out is message is gone ,why the call still fails? Do an iax2 debug, set verbose 5 and check in logs. -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_blutooth
On Fri, Dec 02, 2005 at 08:24:58PM -0500, Jerry Geis wrote: hcitool cc MACHEADSET hcitool auth MACHEADSET hcitool dc MACHEADSET rfcomm bind rfcomm0 MACHEADSET sdptool search --bdaddr MACHEADSET 0x111E These Steps are not necessary, since chan_bluetooth does this for you. however you really should use hcitool browse to find out the right channel for services HS. chan_bluetooth/chan_bluetooth.c:685 sco_thread: wrote 48 to sco Wrong mac or channel in chan_bluetooth I'd say. -- http://www.ukeer.de/about.html If 50 million people say a foolish thing, it is still a foolish thing. --Anatole France ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't compile chan_zap.c on fresh cvs 1.20 checkout
On Saturday 03 December 2005 04:09, Remco Barendse wrote: chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug' chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this chan_zap.c:10870: warning: passing arg 1 of `pri_set_error' from chan_zap.c:10871: warning: passing arg 1 of `pri_set_message' from PRI is involved in every one of these messages... I'd start looking to see if you've got libpri installed, including the libpri headers. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't compile chan_zap.c on fresh cvs 1.20 checkout
On Sat, 3 Dec 2005, Andrew Kohlsmith wrote: On Saturday 03 December 2005 04:09, Remco Barendse wrote: chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug' chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this chan_zap.c:10870: warning: passing arg 1 of `pri_set_error' from chan_zap.c:10871: warning: passing arg 1 of `pri_set_message' from PRI is involved in every one of these messages... I'd start looking to see if you've got libpri installed, including the libpri headers. :-) Hmmm, guess you are right. But this is a home PBX, I'm never going to need a PRI here and in the past libpri was never a requirement or dependency for any asterisk installation. Has this now changed or do I (somewhere, someplace) have some stuff in a config file which make(s) :) asterisk believe it should do something with PRI stuff? Thanks!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rates for Asian countries
Hello,I am looking to hookup my Asterisk box to a gateway provider. I want the cheapest possible rates with highest reliabilty. Countries I am looking for are 1. Pakistan. 2. India. 3. Hong Kong 4. Singapore 5. ChinaIt does not need to be the same provider for all countries. Like I can have one for Pakistan and one for India and so on. Idea is to have the cheapest rates for all these countries. Like for Pakistan I need less then or close to 12cents canadian per mintute. Anyone has any ideas? please pass them on to me. I am looking for options to setup my little dummy calling card setup. Thank you all in advance.Regards, Amir Aziz__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't compile chan_zap.c on fresh cvs 1.20 checkout
On Saturday 03 December 2005 04:09, Remco Barendse wrote: chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug' chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this chan_zap.c:10870: warning: passing arg 1 of `pri_set_error' from chan_zap.c:10871: warning: passing arg 1 of `pri_set_message' from PRI is involved in every one of these messages... I'd start looking to see if you've got libpri installed, including the libpri headers. :-) Hmmm, guess you are right. But this is a home PBX, I'm never going to need a PRI here and in the past libpri was never a requirement or dependency for any asterisk installation. Has this now changed or do I (somewhere, someplace) have some stuff in a config file which make(s) :) asterisk believe it should do something with PRI stuff? You might review each statement in your zapata.conf file to ensure those that are used pertain to whatever card/interface you are using. In past sample configs that folks have posted, some have included things like 'switchtype=national' for their x100p/tdm card, and that parameter (as one example only) is associated with PRI's. For the past two years, I've always compiled and installed zaptel, libpri, and asterisk at the same time. The libpri modules only get loaded _if_ a card is detected in the system that requires them. Certainly doesn't hurt anything. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No CID Info an TE405P with zaptel 1.2.0
Hi, does anyone has experiance with connecting a PRI line to a TE405P card together with zaptel 1.2.0? We are located in Germany and there is nos CID number for incoming calls. What are the right settings for zaptel.conf and zapata.conf? Thanks and regards, bk ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WG: App_rxfax problem
I had the same problem, I am using Asterisk 1.2. I installed spandsp 0.0.3pre6 on the machine then I installed spandsp-0.0.2pre21c I received the same error fax_set_phase_d_handler. It seems I didn't get all of the files associated with 0.0.3pre6 removed before I installed 0.0.2pre21c. To fix the problem I did a make uninstall and manually deleted the spandsp directory in /usr/local/include then reinstalled 0.0.2pre21c. René Enskat [Teamware GmbH] wrote: Dunno :) what do you thing is wrong there? the compile was fine! I only need a solution how to fix this error!! On Sat, 03 Dec 2005 01:52:03 +0800 Steve Underwood [EMAIL PROTECTED] wrote: How could a CVS update fix an error you have made during installation? Steve René Enskat [Teamware GmbH] wrote: so is there a solution in the next cvs udpate? *Von:* René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] *Gesendet:* Donnerstag, 1. Dezember 2005 14:47 *An:* 'asterisk-users@lists.digium.com' *Betreff:* WG: App_rxfax problem I just sent the error in full log: Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so]Dec 1 15:01:08 WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so failed! *Von:* René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] *Gesendet:* Donnerstag, 1. Dezember 2005 08:35 *An:* 'asterisk-users@lists.digium.com' *Betreff:* App_rxfax problem When i load the fax modules into the asterisk i got this errors but compile was ok! I have the latest cvs head [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fwd: Queue Statistics
Hi everyone ! I'm testing Queue Statistics 0.6 from AsteriskGuru Tools with [EMAIL PROTECTED] 2.0, and i gota problem; every call is registered as a new agent when i have configured Static Agents in a Queue, because of Local/[EMAIL PROTECTED] I changed to dynamic agents, and it worked only for the X-Ten Lite Sip Phone being always registered as SIP/101. But when i login as an agent from a GrandStream 101 and a Linksys PAP2-NA, the calls are still being registered as Local/[EMAIL PROTECTED]. I don't have static agents configured in [EMAIL PROTECTED] Any suggestions? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't compile chan_zap.c on fresh cvs 1.20 checkout
On Sat, 3 Dec 2005, Rich Adamson wrote: On Saturday 03 December 2005 04:09, Remco Barendse wrote: chan_zap.c:9318: warning: implicit declaration of function `pri_get_debug' chan_zap.c:10602: error: `PRI_SWITCH_QSIG' undeclared (first use in this chan_zap.c:10870: warning: passing arg 1 of `pri_set_error' from chan_zap.c:10871: warning: passing arg 1 of `pri_set_message' from PRI is involved in every one of these messages... I'd start looking to see if you've got libpri installed, including the libpri headers. :-) Hmmm, guess you are right. But this is a home PBX, I'm never going to need a PRI here and in the past libpri was never a requirement or dependency for any asterisk installation. Has this now changed or do I (somewhere, someplace) have some stuff in a config file which make(s) :) asterisk believe it should do something with PRI stuff? You might review each statement in your zapata.conf file to ensure those that are used pertain to whatever card/interface you are using. In past sample configs that folks have posted, some have included things like 'switchtype=national' for their x100p/tdm card, and that parameter (as one example only) is associated with PRI's. For the past two years, I've always compiled and installed zaptel, libpri, and asterisk at the same time. The libpri modules only get loaded _if_ a card is detected in the system that requires them. Certainly doesn't hurt anything. It worked. Strange, but true. I checked through all my configs and there is nothing there that even remotely hints to a PRI. Thanks for the solution :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZapHFC cards not maintaining sync?!
On Fri, December 2, 2005 22:54, Francesco Peeters said: On Fri, December 2, 2005 22:50, Francesco Peeters said: On Fri, December 2, 2005 21:45, Kristof Hardy said: Francesco Peeters wrote: Does anybody have any experience in this? I am using * 1.2 BRIstuffed 0.3.0 Pre1 No experience on that, but there's an updated bristuff (0.3.0pre1b), maybe try that one? This is 1 issue that's fixed: - chan_zap/libpri fixes (stuck B channels) Just installed 0.3.0pre1c, but no change! :-/ I have now got this little ditty running to keep an eye on it: while true; do grep (F /proc/zaptel/2; sleep .1; done I do see the once a minute down-time come by as a combination of 1 F4, 2 F6's and then F7's. When it goes down for an extended time, it shows 1 F4 and a lot of F6's before finally returning F7's again... :-( Watching the console for a while I see regular messages, which I could also find in /var/log/messages: Dec 3 16:37:15 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: empty HDLC frame received. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Can anyone with one or more HFC-PCI card(s) (esp. in The Netherlands) check if they see these on a regular basis as well? (And I am talking many times an hour here!) TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour
I just upgraded my config from * 1.0.10 to 1.2 I removed caller ID from my configs because when I try to use CallerID (new style) on my IAX provider (magrathea) but whenever I try to make a call I get a message from the provider that You are not registered to use this service. Removing the callerid stuff seems to solve this. I guess they are not ready for the new updated IAX protocol? Anyways, now to my real problem. I have a TDM11B card. Obviously one connection to the phone line, one connection to an analog phone. I just used the exact same config files as with * 1.0.10 I have this in my /etc/asterisk/zapata.conf: callerid=202 signalling=fxo_ks group=1 context=intern-all channel = 1 Whenever I pick up that phone I get on the console: Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Okay, but I want to make an OUTGOING call so I don't need this phone to be in default context, do I?? When I add the default context (with s extension) to intern-all whenever I pick up the analog phone it starts ringing my default context like a bat phone. Nice but not what I wanted.. I just want it to give me a dial tone and wait for the number I want to dial. What am I overlooking here?? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma Asterisk at home
On Dec 2, 2005, at 12:03 PM, Jess Coburn wrote: Guys, I'm curious if it's possible to asterisk at home and the sangoma T1 cards together. I realize asteriskathome is traditionally used for at home, but I'd like to use it in a small office with a T1 and our hardware is a Sangoma card. I know all I need to do to get the sangoma working is recompile the zaptel but I can't seem to find the source, etc on the server after asteriskathome installs. Jess Jess, Log on to [EMAIL PROTECTED] and type help-aah. It will return a list of commands you can use to configure your [EMAIL PROTECTED] system. (You did already do this and change all of the passwords, right?) One of those commands is rebuild_zaptel which will rebuild the zaptel drivers for you. You will also need to do this every time a new kernel is installed when you update the server using yum. You might find it helpful to go to asteriskathome.sourceforge.net and spend some time perusing the handbook. Tom PS: If you haven't changed the passwords on your system, do it now!! Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_blutooth
I have had the same issue. The headset connects, but there is no audio. I have a IOGEar usb dongle. However, if i use my Q-Stor usb bluetooth dongle I do have audio. Doing a sdptool browse for the rfcomm channel. it is the same in both cases. So it's something with the dongle not accepting some of the commands. Also, with the IOGear. I get errors do not know how to condition -1 Ben On Fri, Dec 02, 2005 at 08:24:58PM -0500, Jerry Geis wrote: hcitool cc MACHEADSET hcitool auth MACHEADSET hcitool dc MACHEADSET rfcomm bind rfcomm0 MACHEADSET sdptool search --bdaddr MACHEADSET 0x111E These Steps are not necessary, since chan_bluetooth does this for you. however you really should use hcitool browse to find out the right channel for services HS. chan_bluetooth/chan_bluetooth.c:685 sco_thread: wrote 48 to sco Wrong mac or channel in chan_bluetooth I'd say. -- http://www.ukeer.de/about.html If 50 million people say a foolish thing, it is still a foolish thing. --Anatole France ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour
On Sat, 3 Dec 2005, Remco Barende wrote: Whenever I pick up that phone I get on the console: Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Have you by chance set immediate to yes? IIRC, there's a feature that will send you to the configured context as soon as you pick up your phone (this is in zapata.conf). Might be worth checking that out. Cheers, Gerald. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo!!
Hello again i'm very new with this theme i'm testing the [EMAIL PROTECTED] 2.1 recently i donwloaded from the sourceforge.net i need info about the intel chipset modem to call and receiving calls and the configruration for internal extensionsl work ok, i install in 3 computers the xten free softphone and work ok but have too much echo, how i do to minimize or delete the echo sound?? and the other question is how i do to install the spanish voices? i va 2 spanish voices to install in the system but i dont have idea on how to do this because i log in the server with the root user but in the /mnt directory dont have access to the cdrom disc the same problem is in the /media directory Thanks in advance to all!!! Vladimir __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma Asterisk at home
Thanks for the feedback guys. yeah I RTFM'd last time but honestly wanted the box up quickly so I gave up too early. Today, I'm reloading it now. I have the 2.1 ISO installing but I read in their forums there's possibly a problem with the install that's on the Georgia mirror and that the UK mirror was good so I'm downloading that now. sounds like maybe the gcc-c++ was what I was missing. We'll know in about 30 minutes. Glad to know it works and thanks again for the help, I'll keep you posted as I progress Jess PS yes no default passwords over here. On 12/3/05, Tom Rymes [EMAIL PROTECTED] wrote: On Dec 2, 2005, at 12:03 PM, Jess Coburn wrote: Guys, I'm curious if it's possible to asterisk at home and the sangoma T1 cards together. I realize asteriskathome is traditionally used for at home, but I'd like to use it in a small office with a T1 and our hardware is a Sangoma card. I know all I need to do to get the sangoma working is recompile the zaptel but I can't seem to find the source, etc on the server after asteriskathome installs. JessJess,Log on to [EMAIL PROTECTED] and type help-aah. It will return a list of commands you can use to configure your [EMAIL PROTECTED] system. (You did already do thisand change all of the passwords, right?) One of those commands isrebuild_zaptel which will rebuild the zaptel drivers for you. You will also need to do this every time a new kernel is installedwhen you update the server using yum.You might find it helpful to go to asteriskathome.sourceforge.net andspend some time perusing the handbook.TomPS: If you haven't changed the passwords on your system, do it now!!Tom RymesCascade Link Systems www.cascadelinksystems.com(603) 375-1414Intelligent technology solutions for small businesses.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZapHFC cards not maintaining sync?!
On Sat, December 3, 2005 16:40, Francesco Peeters (Asterisk) said: On Fri, December 2, 2005 22:54, Francesco Peeters said: Watching the console for a while I see regular messages, which I could also find in /var/log/messages: Dec 3 16:37:15 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: empty HDLC frame received. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Can anyone with one or more HFC-PCI card(s) (esp. in The Netherlands) check if they see these on a regular basis as well? (And I am talking many times an hour here!) TIA! Ok, I have been analyzing the activities and see the following: 1) Every 10 seconds () the D channel gets torn down, which 2) Results in the CRC error, which means that 3) Every 3 minutes, the D channel goes down for EXACTLY 1 minute. This means there is a 66% chance of actually being able to use the ISDN link, and thus use it to dial out or be dialed on... This is obviously not acceptable for a PBX... I could try to get the KPN to give me a permanent D channel, but are there any tricks to try that would/could make asterisk somehow keep up the D channel?... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour
On Sat, 3 Dec 2005, Begumisa Gerald M wrote: On Sat, 3 Dec 2005, Remco Barende wrote: Whenever I pick up that phone I get on the console: Dec 3 16:37:36 WARNING[19551]: pbx.c:2347 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Zap/1-1' Have you by chance set immediate to yes? IIRC, there's a feature that will send you to the configured context as soon as you pick up your phone (this is in zapata.conf). Might be worth checking that out. I have but only for the phone line, it is immediately after: signalling=fxs_ks immediate=yes I did some further testing, this happens only after I have done a RELOAD on the console. When I exit asterisk and start asterisk again all seems to be working as normal. Maybe it's an * bug? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZapHFC cards not maintaining sync?!
On Sat, 3 Dec 2005, Francesco Peeters (Asterisk) wrote: On Sat, December 3, 2005 16:40, Francesco Peeters (Asterisk) said: On Fri, December 2, 2005 22:54, Francesco Peeters said: Watching the console for a while I see regular messages, which I could also find in /var/log/messages: Dec 3 16:37:15 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: empty HDLC frame received. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. This is not normal. Run the florz patch over your bristuff install (I'm assuming you are using bristuff). These problems will cause your box to hang after anything beteen 5 and 48 hours. Can anyone with one or more HFC-PCI card(s) (esp. in The Netherlands) check if they see these on a regular basis as well? (And I am talking many times an hour here!) I am in NL :) 1) Every 10 seconds () the D channel gets torn down, which That's too slow, it should happen about every 1-2 seconds or so. The d channel going down and up again is normal behaviour. 2) Results in the CRC error, which means that 3) Every 3 minutes, the D channel goes down for EXACTLY 1 minute. I could try to get the KPN to give me a permanent D channel, but are there any tricks to try that would/could make asterisk somehow keep up the D channel?... Good luck with our Royal Dutch KPN, but I would try florz first :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 and weird ZAP interface behaviour
On Sat, 3 Dec 2005, Remco Barende wrote: I have but only for the phone line, it is immediately after: signalling=fxs_ks immediate=yes What I actually meant is that you should turn this off if you don't need the functionality. Most likely you are defining the extension channel after the phone line thus it is inheriting the setting as well. Gerald. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys SPA-841 Missing Calls
I experienced a similar situation with the SPA-841, it turned out to be that the calls I was missing didn't have caller ID (outside calls with caller ID Blocked), found that the SPA841 phone has an option to ignore calls without caller ID. Turned this option off and it fixed the problem. Sorry, I no longer use the SPA841 and I can't remember the exact menu setting on the SPA841 that fixed it, so you will have to go through the manual. c Message: 1 Date: Fri, 02 Dec 2005 21:43:01 -0800 From: Wolfgang S. Rupprecht [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Linksys SPA-841 Missing Calls To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Might the SPA-841 be crashing and rebooting? With the current firmware (v. 3.1.4) I often see my phone hang and flash all its lights Really? For me the 841 is a quite stable phone. Out of the 15 we have in the office neither one crashed in the past 3 months. And they are used heavily. The phone has weaknesses, but stability in my opinion is not one of them. Phone info: Software Version: 3.1.4(a) Hardware Version: 1.0.0(1813) Elapsed Time: 50 days and 09:48:10 I only have 1 phone so it is hard to tell if the crashing is a hardware or software problem. I never noticed the phone having problems previous to this. I did resync asterisk to HEAD a month ago. Thats also about the time the phone started crashing (or at least I started noticing it). Come to think of it, I've been running the current firmware in the phone since July 20th. The only think that changed in recently was asterisk. I wonder if there is something the newer asterisk is doing that the phone really hates... Asterisk CVS HEAD built by [EMAIL PROTECTED] on a amd64 running OpenBSD on 2005-11-02 00:58:42 UTC Software Version: 3.1.4(a) Hardware Version: 1.0.0(700b) Elapsed Time: 1 day and 05:54:03 (crashed during a call) People have been reporting a finicky ethernet connector, so maybe that is the reason the phone does not answer to any traffic? Yea, this phone has that problem too. ;-) Some cables just don't work. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I escape queue with a '*'?
Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension Manual
in wath link or page is the * commands for the phone extensions?? example *79 is for on or off the extension ?? Thanks again in advance - Original Message - From: Chuck Bunn [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:06 PM Subject: [Asterisk-Users] Can I escape queue with a '*'? Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX Conf Realtime?
Im in the same situation, I have been trying to google around a lot for some examples but havent come up with much. Hi I've been playing around with Realtime Asterisk using the ODBCmodule to connect to my database and I got extensions working but nowI'm looking to get my iax.conf into the database. I would like to have the users who can register with my box to dial extensions in there,and also the connections to my outbound providers (voicepulse, voxee,etc etc). I've been trying to read the wiki but it doesnt really havea good documentation of what I need to do with my current iax.conf totell it to look in the database for things. Do I have to use switch =Realtime like I do in extensions.conf?Any help or information here is appreciated. Thanks alot! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Order of ports on rear of Sangoma card and pictures in a mini-itx chassis.
I'm getting much nearer in getting my Sangoma A2022-SO analog card working with Asterisk 1.2, however I am unsure of the ordering of ports on the rear of the card. I've taken some pictures of the card in the hope anyone can help me guess which physical ports relate to the 2 x fxs and 2 x fxo ports. http://www.flickr.com/photos/mikedent/69777387/in/set-1494048/ You can also see a couple of pictures of this nice card installed in a mini-itx chassis, could make a nice small scale PBX when I get it all working :) ! http://www.flickr.com/photos/mikedent/sets/1494048/ Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax2 connection failed
Hi: i made the debug and look what i get: dial [EMAIL PROTECTED] -- Executing Dial(OSS/dsp, iax2/callshopcompany/0017046872001) in new stack -- Called callshopcompany/0017046872001 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00016ms SCall: 1 DCall: 0 [213.61.187.150:4569] VERSION : 2 CALLED NUMBER : 0017046872001 LANGUAGE: en USERNAME: XX FORMAT : 256 CAPABILITY : 63744 ADSICPE : 0 DATE TIME : 193180416 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00025ms SCall: 00061 DCall: 1 [213.61.187.150:4569] AUTHMETHODS : 3 CHALLENGE : 150845023 USERNAME: XXX Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00274ms SCall: 1 DCall: 00061 [213.61.187.150:4569] MD5 RESULT : 3a301d3fc9e1e38504ab6d366c5740f5 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00274ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00256ms SCall: 00061 DCall: 1 [213.61.187.150:4569] FORMAT : 256 -- Call accepted by 213.61.187.150 (format g729) -- Format for call is g729 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00256ms SCall: 1 DCall: 00061 [213.61.187.150:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10017ms SCall: 1 DCall: 00061 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10017ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 10017ms SCall: 1 DCall: 00061 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10022ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10022ms SCall: 1 DCall: 00061 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10022ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: HANGUP Timestamp: 11029ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Unknown IE 042 : Present Ignoring unknown information element 'Unknown IE' (42) of length 1 Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 11029ms SCall: 1 DCall: 00061 [213.61.187.150:4569] -- Hungup 'IAX2/callshopcompany/1' == No one is available to answer at this time -- Executing Hangup(OSS/dsp, ) in new stack --- Administrator TOOTAI [EMAIL PROTECTED] wrote: chawki hammoud a écrit : Hi: Now the time out is message is gone ,why the call still fails? Do an iax2 debug, set verbose 5 and check in logs. -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Manual
*411 Directory *43 Echo Test *60 Time *61 Weather *62 Schedule wakeup call *65 festival test (your extension is XXX) *70 Activate Call Waiting (deactivated by default) *71 Deactivate Call Waiting *72 Call Forwarding System *73 Disable Call Forwarding *77 IVR Recording *78 Enable Do-Not-Disturb *79 Disable Do-Not-Disturb *90 Call Forward on Busy *91 Disable Call Forward on Busy *97 Message Center (does no ask for extension) *98 Enter Message Center *99 Playback IVR Recording *666 Test Fax Simulate incoming call - Original Message - From: Vladimir Montealegre [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:34 PM Subject: [Asterisk-Users] Extension Manual in wath link or page is the * commands for the phone extensions?? example *79 is for on or off the extension ?? Thanks again in advance - Original Message - From: Chuck Bunn [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:06 PM Subject: [Asterisk-Users] Can I escape queue with a '*'? Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Manual
what is your question? you must set up extensions yourself in extensions.conf...you can set the extensions to whatever you want (say, if you are replacing an existing PBX and want the users to have the same extensions). -yair On 12/3/05, Vladimir Montealegre [EMAIL PROTECTED] wrote: *411 Directory *43 Echo Test *60 Time *61 Weather *62 Schedule wakeup call *65 festival test (your extension is XXX) *70 Activate Call Waiting (deactivated by default) *71 Deactivate Call Waiting *72 Call Forwarding System *73 Disable Call Forwarding *77 IVR Recording *78 Enable Do-Not-Disturb *79 Disable Do-Not-Disturb *90 Call Forward on Busy *91 Disable Call Forward on Busy *97 Message Center (does no ask for extension) *98 Enter Message Center *99 Playback IVR Recording *666 Test Fax Simulate incoming call - Original Message - From: Vladimir Montealegre [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:34 PM Subject: [Asterisk-Users] Extension Manual in wath link or page is the * commands for the phone extensions?? example *79 is for on or off the extension ?? Thanks again in advance - Original Message - From: Chuck Bunn [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:06 PM Subject: [Asterisk-Users] Can I escape queue with a '*'? Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone have experience with SellVoip.net
Very slow response, if they have the DID you want in stock you get it right away, they dont seem to be able to port toll or DIDs as advertised. They claim to have a server in Washington State and Florida but no centralized server, they are too far for me to consider good service. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 02, 2005 10:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anyone have experience with SellVoip.net Does anyone have any experience with SellVoip.net? Their DID pricing and termination pricing seems pretty good. How is their service? How easy are they to contact should any problems arrise? -jglucky ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZapHFC cards not maintaining sync?!
On Sat, December 3, 2005 19:01, Remco Barende said: On Sat, 3 Dec 2005, Francesco Peeters (Asterisk) wrote: On Sat, December 3, 2005 16:40, Francesco Peeters (Asterisk) said: On Fri, December 2, 2005 22:54, Francesco Peeters said: Watching the console for a while I see regular messages, which I could also find in /var/log/messages: Dec 3 16:37:15 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: empty HDLC frame received. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. This is not normal. Run the florz patch over your bristuff install (I'm assuming you are using bristuff). These problems will cause your box to hang after anything beteen 5 and 48 hours. Already HAVE Florz patch installed! :-( What version of * and BRIstuff are you using? Can anyone with one or more HFC-PCI card(s) (esp. in The Netherlands) check if they see these on a regular basis as well? (And I am talking many times an hour here!) I am in NL :) I assumed as much when I saw your last name... :-) Whereabouts in NL? I'm in Zoetermeer (ZH)... 1) Every 10 seconds () the D channel gets torn down, which That's too slow, it should happen about every 1-2 seconds or so. The d channel going down and up again is normal behaviour. I know it is. Used to work for a Networking Competence Centre, and we had the same kind of issues with 3Com Netbuilders. The first call attempt after the D Channel was torn down always failed... The only solution was to get KPN to turn on the D Channel permanently... 2) Results in the CRC error, which means that 3) Every 3 minutes, the D channel goes down for EXACTLY 1 minute. I could try to get the KPN to give me a permanent D channel, but are there any tricks to try that would/could make asterisk somehow keep up the D channel?... I noticed that the 'deactivated' issue doesn't happen for a while after a call has been placed. I am now testing placing a call every minute, with a 100 ms timeout using the manager api. This means it never actually gets a chance to get through, or be picked up, but it does cause activity on the D channel. This has been running for half an hour now, and I haven't seen the channel go down for extended periods since. I'm not sure whether the KPN will like it, but it's an interesting test to run! G Good luck with our Royal Dutch KPN, but I would try florz first :) Tell me about it! Like I said above, we had *extensive* experience with them over the D Channel issue! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax2 connection failed
On 3 Dec 2005, at 20:27, chawki hammoud wrote:Hi:i made the debug and look what i get: dial [EMAIL PROTECTED] -- Executing Dial("OSS/dsp","iax2/callshopcompany/0017046872001") in new stack -- Called callshopcompany/0017046872001Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type:IAX Subclass: NEW Timestamp: 00016ms SCall: 1 DCall: 0[213.61.187.150:4569] VERSION : 2 CALLED NUMBER : 0017046872001 LANGUAGE : en USERNAME : XX FORMAT : 256 CAPABILITY : 63744 ADSICPE : 0 DATE TIME : 193180416Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type:IAX Subclass: ACK Timestamp: 00016ms SCall: 00061 DCall: 1[213.61.187.150:4569]Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type:IAX Subclass: AUTHREQ Timestamp: 00025ms SCall: 00061 DCall: 1[213.61.187.150:4569] AUTHMETHODS : 3 CHALLENGE : 150845023 USERNAME : XXXTx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type:IAX Subclass: AUTHREP Timestamp: 00274ms SCall: 1 DCall: 00061[213.61.187.150:4569] MD5 RESULT : 3a301d3fc9e1e38504ab6d366c5740f5Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type:IAX Subclass: ACK Timestamp: 00274ms SCall: 00061 DCall: 1[213.61.187.150:4569]Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type:IAX Subclass: ACCEPT Timestamp: 00256ms SCall: 00061 DCall: 1[213.61.187.150:4569] FORMAT : 256 -- Call accepted by 213.61.187.150 (format g729) -- Format for call is g729Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type:IAX Subclass: ACK Timestamp: 00256ms SCall: 1 DCall: 00061[213.61.187.150:4569]Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type:IAX Subclass: LAGRQ Timestamp: 10017ms SCall: 1 DCall: 00061[213.61.187.150:4569]Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type:IAX Subclass: LAGRP Timestamp: 10017ms SCall: 00061 DCall: 1[213.61.187.150:4569]Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type:IAX Subclass: ACK Timestamp: 10017ms SCall: 1 DCall: 00061[213.61.187.150:4569]Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type:IAX Subclass: LAGRQ Timestamp: 10022ms SCall: 00061 DCall: 1[213.61.187.150:4569]Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type:IAX Subclass: LAGRP Timestamp: 10022ms SCall: 1 DCall: 00061[213.61.187.150:4569]Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type:IAX Subclass: ACK Timestamp: 10022ms SCall: 00061 DCall: 1[213.61.187.150:4569]Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type:IAX Subclass: HANGUP Timestamp: 11029ms SCall: 00061 DCall: 1[213.61.187.150:4569] Unknown IE 042 : PresentIgnoring unknown information element 'Unknown IE' (42)of length 1Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type:IAX Subclass: ACK Timestamp: 11029ms SCall: 1 DCall: 00061[213.61.187.150:4569] -- Hungup 'IAX2/callshopcompany/1' == No one is available to answer at this time -- Executing Hangup("OSS/dsp", "") in new stack--- Administrator TOOTAI [EMAIL PROTECTED] wrote:Well, I see 3 things here: 1) your asterisk is _only_ prepared to talk 729 - nothing else - is that what you want? 2) they accept the call then hang up before sending any voice data (or any RINGBACK) 3) your asterisk isn't understanding the reason they are giving for hanging upThe last one is a bit of a puzzle - is this a very old asterisk install? (or is the other end old?)From the documentation, IE 042 is CauseCode , just what you would like to know!If it were me I'd a) allow some other codecs to see if that's the problem b) ethereal the traffic and see what the cause code value ispossible values are: | 1 | Unassigned/unallocated number | | 2 | No route to specified transit network | | 3 | No route to destination | | 6 | Channel unacceptable | | 7 | Call awarded and delivered | | 16 | Normal call clearing | | 17 | User busy | | 18 | No user response | | 19 | No answer | | 21 | Call rejected | | 22 | Number changed | | 27 | Destination out of order | | 28 | Invalid number format/incomplete number | | 29 | Facility rejected | Good luck,Tim. http://www.westhawk.co.uk/
Re: [Asterisk-Users] Iax2 connection failed
Hi: Thanks for your answer, i tried all possible codecs and the same result the call failed,my asterisk verison is 1.0 ,I asked callshopcompany the voip provider about whats the reason of the failure of the calls and he said he didnt know whats the problem and he's all customers making succesful calls to their iax server without any problems. NOTICE: i make successful calls through sip to the same voip provider. --- tim panton [EMAIL PROTECTED] wrote: On 3 Dec 2005, at 20:27, chawki hammoud wrote: Hi: i made the debug and look what i get: dial [EMAIL PROTECTED] -- Executing Dial(OSS/dsp, iax2/callshopcompany/0017046872001) in new stack -- Called callshopcompany/0017046872001 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00016ms SCall: 1 DCall: 0 [213.61.187.150:4569] VERSION : 2 CALLED NUMBER : 0017046872001 LANGUAGE: en USERNAME: XX FORMAT : 256 CAPABILITY : 63744 ADSICPE : 0 DATE TIME : 193180416 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00025ms SCall: 00061 DCall: 1 [213.61.187.150:4569] AUTHMETHODS : 3 CHALLENGE : 150845023 USERNAME: XXX Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00274ms SCall: 1 DCall: 00061 [213.61.187.150:4569] MD5 RESULT : 3a301d3fc9e1e38504ab6d366c5740f5 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00274ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00256ms SCall: 00061 DCall: 1 [213.61.187.150:4569] FORMAT : 256 -- Call accepted by 213.61.187.150 (format g729) -- Format for call is g729 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00256ms SCall: 1 DCall: 00061 [213.61.187.150:4569] Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: LAGRQ Timestamp: 10017ms SCall: 1 DCall: 00061 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10017ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 10017ms SCall: 1 DCall: 00061 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10022ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: LAGRP Timestamp: 10022ms SCall: 1 DCall: 00061 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10022ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: HANGUP Timestamp: 11029ms SCall: 00061 DCall: 1 [213.61.187.150:4569] Unknown IE 042 : Present Ignoring unknown information element 'Unknown IE' (42) of length 1 Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 11029ms SCall: 1 DCall: 00061 [213.61.187.150:4569] -- Hungup 'IAX2/callshopcompany/1' == No one is available to answer at this time -- Executing Hangup(OSS/dsp, ) in new stack --- Administrator TOOTAI [EMAIL PROTECTED] wrote: Well, I see 3 things here: 1) your asterisk is _only_ prepared to talk 729 - nothing else - is that what you want? 2) they accept the call then hang up before sending any voice data (or any RINGBACK) 3) your asterisk isn't understanding the reason they are giving for hanging up The last one is a bit of a puzzle - is this a very old asterisk install? (or is the other end old?) From the documentation, IE 042 is CauseCode , just what you would like to know! If it were me I'd a) allow some other codecs to see if that's the problem b) ethereal the traffic and see what the cause code value is possible values are: |1| Unassigned/unallocated number| |2| No route to specified transit network| |3| No route to destination | |6| Channel unacceptable | |7| Call awarded and delivered
[Asterisk-Users] Call queues, agents with DND status set.
-- Called 1101 -- Agent/1101 is ringing -- Got SIP response 480 Temporarily Unavailable back from x.x.x.x -- SIP/1101-9b08 is circuit-busy Is it possible to force logoff such agents? -- Vladimir S. Blazhkun, Personal Communications Systems, LLC. Leading IP NCC Specialist, Work phone: +7 095 7847617. JNCIA-M #773, JNCIS-M #1100. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Manual
Hi, I understand that I must set up extensions myself but in the release notes for Asterisk 1.2 it specifically states that multiple digit extensions can be used in the exit context of a queue. Prior to version 1.2 only a single digit would would when exiting from a queue. I have tried setting up the '*' in a exit context for a queue but it does not seem to work. I can get a single digit to work without a problem. I have not tried multiple digits. I figured since multiple digits worked that perhaps the *' or '#' might work as well... Thanks Yair Hakak wrote: what is your question? you must set up extensions yourself in extensions.conf...you can set the extensions to whatever you want (say, if you are replacing an existing PBX and want the users to have the same extensions). -yair On 12/3/05, Vladimir Montealegre [EMAIL PROTECTED] wrote: *411 Directory *43 Echo Test *60 Time *61 Weather *62 Schedule wakeup call *65 festival test (your extension is XXX) *70 Activate Call Waiting (deactivated by default) *71 Deactivate Call Waiting *72 Call Forwarding System *73 Disable Call Forwarding *77 IVR Recording *78 Enable Do-Not-Disturb *79 Disable Do-Not-Disturb *90 Call Forward on Busy *91 Disable Call Forward on Busy *97 Message Center (does no ask for extension) *98 Enter Message Center *99 Playback IVR Recording *666 Test Fax Simulate incoming call - Original Message - From: Vladimir Montealegre [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:34 PM Subject: [Asterisk-Users] Extension Manual in wath link or page is the * commands for the phone extensions?? example *79 is for on or off the extension ?? Thanks again in advance - Original Message - From: Chuck Bunn [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:06 PM Subject: [Asterisk-Users] Can I escape queue with a '*'? Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma Asterisk at home
I just wanted to say thanks to everything that provided feedback and assistance on this. I have AAH2.1 running with my Sangoma T1 card. Here's a few of the gotchas I ran into and my process, please note I'm not an asterisk expert and this could be totally wrong but here's what worked for me: 1. you have to compile and install the sangoma files, there's a doc wiki at sangoma.editme.com that should help 2. you have to remove ztdummy from the start up modules. I don't know the most elegant way to do this but I edited /etc/init.d/zaptel and commented out the line that loads it 3. you have to configure the sangoma card using wancfg (very important) 4. edit the etc/zaptel.conf and etc/asterisk/zapata.conf files That's it so far... Jess ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adit 600 and Groundstart
Doug Lytle wrote: Hey everybody. I have an Adit 600 that I'm not able to get working properly with Groundstart. The Adit (BootCode 1.23), 1 TDM (Version 6.1.2) and 1 FXO card (Version 1.12). For the sake of completeness and the archives, the answer to this problem was discovered in the Asterisk 'The Future of Telephony book. The zconfig.h file in the zaptel directory needed to be edited and the Enable CAC ground start signaling needed to be uncommented. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 problems
I have similar problems with call drops.I don't know if "Shadow Ping" is some kind of pinging software. I have run a lot of flood pinging and everything comes back just fine. I don't have Cisco phones, I use Softphones and it's the only application running on the PCs (aside from MS Windows and whatever it runs in the background).I notice the problem mainly on the softphones. I also have some Uniden hardphones that don't seem to present the same problems. The softphones are eyeBeam. I wonder if it's a bug in the softphone or simply that the machines could be infected with some virus or something like that. It happens randomly (with a lot of calls or few calls, some PCs yes and some PCs no).Could this still be network related?Thanks,Waldo On Dec 2, 2005, at 5:17 PM, jonc wrote:I do run Ethereal on mine when looking for real-time problems. It's great for helping you see what is going on at the packet level, but it is the wrong tool for measuring Latency/QOS problems. Shadow Ping works fairly well for measuring latencies. In earlier times I used to just run a quasi-flood ping to the offending phone (10 pings/second) and look for latency variations and dropped packets. On a clean network with no problems there should be NO dropped packets, and latency variations should be minimal. You'll find some interesting problems that accompany the use of cheap unmanaged switches (and please don't tell me you are trying to use hubs!). For our setups we use either Cisco 2900XL-EN or Cisco 3500 series switches. This come with built in VoIP detection at the port level *and* allow us to use VLANs to separate out Voice and Data. They are champion workhorses and using them lets us also run single wire to the desktop (running the PC off the pass-thru switch on the back of the phone). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Converted mp3 files to raw for musiconhold and still does not work...
Hi, I setup music on hold as directed for Asterisk 1.2 but still no music on hold. Any ideas what I did wrong. I see it start in the CLI but then it immediately stops?? I also see the Hangup occur 20 seconds later as it should according to WitMusicOnHold(20). I used a test setup suggested in the wiki... ** CLI Output Spawn extension (longdistance, 870, 4) exited non-zero on 'SIP/499-39ff' -- Executing Answer(SIP/499-206c, ) in new stack -- Executing SetMusicOnHold(SIP/499-206c, default) in new stack -- Executing WaitMusicOnHold(SIP/499-206c, 20) in new stack -- Started music on hold, class 'default', on channel 'SIP/499-206c' -- Stopped music on hold on SIP/499-206c == Spawn extension (longdistance, 870, 3) exited non-zero on 'SIP/499-206c' *** ;Music on Hold exten = 870,1,Answer exten = 870,2,SetMusicOnHold(default) exten = 870,3,WaitMusicOnHold(20) exten = 870,4,Hangup * musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 * zapata.conf [trunkgroups] [channels] musiconhold=default echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=10.0 txgain=3.0 usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=no transfer=no immediate=no faxdetect=both context=incoming-home signalling=fxs_ks group=1 channel = 1,2 context=local signalling=fxo_ks group=2 channel = 3 context=longdistance signalling=fxo_ks group=3 channel = 4 ** Output of /var/lib/asterisk/mohmp3 directory [EMAIL PROTECTED] mohmp3]# ls -la total 107772 drwxr-xr-x 2 asterisk asterisk 4096 Dec 3 20:33 . drwxr-xr-x 9 asterisk asterisk 4096 Nov 11 10:18 .. -rw-r--r-- 1 root root 1939812 Nov 11 10:24 fpm-calm-river.mp3 -rw-r--r-- 1 root root 2582496 Nov 11 10:24 fpm-sunshine.mp3 -rw-r--r-- 1 root root 2217563 Nov 11 10:24 fpm-world-mix.mp3 -rw-r--r-- 1 asterisk asterisk 884864 Oct 29 12:39 QuajiroPromo.mp3 -rw-r--r-- 1 asterisk asterisk 835712 Oct 29 12:39 TristeAlegriaPromo.mp3 Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Converted mp3 files to raw for musiconhold and still does not work...
I had the same problem, had to reboot the machine it started to work. ThanksOn 12/3/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi,I setup music on hold as directed for Asterisk 1.2 but still no music onhold. Any ideas what I did wrong. I see it start in the CLI but then itimmediately stops?? I also see the Hangup occur 20 seconds later as it should according to WitMusicOnHold(20). I used a test setup suggested inthe wiki...**CLI OutputSpawn extension (longdistance, 870, 4) exited non-zero on 'SIP/499-39ff'-- Executing Answer(SIP/499-206c, ) in new stack -- Executing SetMusicOnHold(SIP/499-206c, default) in new stack-- Executing WaitMusicOnHold(SIP/499-206c, 20) in new stack-- Started music on hold, class 'default', on channel 'SIP/499-206c' -- Stopped music on hold on SIP/499-206c== Spawn extension (longdistance, 870, 3) exited non-zero on'SIP/499-206c'***;Music on Holdexten = 870,1,Answerexten = 870,2,SetMusicOnHold(default) exten = 870,3,WaitMusicOnHold(20)exten = 870,4,Hangup*musiconhold.conf[default]mode=quietmp3directory=/var/lib/asterisk/mohmp3* zapata.conf[trunkgroups][channels]musiconhold=defaultechocancel=yesechocancelwhenbridged=yesechotraining=yesrxgain=10.0txgain=3.0usecallerid=yeshidecallerid=nocallwaiting=no threewaycalling=notransfer=noimmediate=nofaxdetect=bothcontext=incoming-homesignalling=fxs_ksgroup=1channel = 1,2context=localsignalling=fxo_ksgroup=2channel = 3 context=longdistancesignalling=fxo_ksgroup=3channel = 4**Output of /var/lib/asterisk/mohmp3 directory[EMAIL PROTECTED] mohmp3]# ls -latotal 107772drwxr-xr-x2 asterisk asterisk 4096 Dec3 20:33 . drwxr-xr-x9 asterisk asterisk 4096 Nov 11 10:18 ..-rw-r--r--1 root root1939812 Nov 11 10:24 fpm-calm-river.mp3-rw-r--r--1 root root2582496 Nov 11 10:24 fpm-sunshine.mp3-rw-r--r--1 root root2217563 Nov 11 10:24 fpm-world-mix.mp3-rw-r--r--1 asterisk asterisk 884864 Oct 29 12:39 QuajiroPromo.mp3-rw-r--r--1 asterisk asterisk 835712 Oct 29 12:39 TristeAlegriaPromo.mp3Thanks___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom backlight?
I have actually spoken to a Polycom product engineer at VON and brought this up. It seems like it's a frequent request, hopefully they will address it soon. On 12/2/05, Wilson Pickett [EMAIL PROTECTED] wrote: Official Polycom view seems to be that you shouldn't work at night :)The phones are crying out for a backlit LCD that only lights whenambient light is low. I have a cheap radio/weather station with alarge LCD that does that. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk cluster and astdb
This is correct, astdb isn't stored in a database and essentially your scenario with astdb file being shared won't work or will be highly unreliable. I am almost sure there was a patch in circulation, alas I didn't find it in the bug tracker. On 12/1/05, Bruce Ferrell [EMAIL PROTECTED] wrote: Matt Riddell wrote: dashy dude wrote:Dear AllI am trying to build a high availability cluster ofasterisk.I am using RedHat cluster management suit onEnterprise edition AS3 Origianally, astdb was located on native hard disk ofeach server.All my end points are configured for Reinvite=YesEverrything was working fine and if active server is rebooted, the standby would take over and the ongoingcalls will continue without any problem.But this had a problem that the astdb file is notupdated with latest end-point information and phones dont get a call untill they re register.To avoid this, I moved the astdb file on the sharedstorage and created sym links from individual servers.Now, when the active server is rebooted, all the active calls are dropped.Please help me in resolving this. Why don't you use Asterisk RealTime?correct if I'm wrong (frequently) but the call state isn't stored in the realtime db is it?linux-ha uses a form of shared disk called DRBD that might solve this ifyou forced the astdb onto that.Only one node of the cluster iscurrently allowed to write to astdb on that though. Just a thought___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadband VoIP Startup with Asterisk
4) What is some good company names to purchase DID's and VoIP termination from? I have been looking at VoIPJet and Teliax. In my six months of using both services full time, both have worked reliably, but we use Teliax for DIDs as we need a company we can talk to when we need something, and Teliax is wonderful to deal with. I have had to move DIDs between accounts as we brought more servers online, and had to deal with some problems with multiple account son one machine, Teliax have spent time on the phone figuring out what I need and how to make their system do it. The VOIPJet rates are a little better but I only use them as standby as the cheapest rates are not that relevant to me, good support isn't free and I'm prepared to pay a little extra for it. The Terms of Service are a huge turn off, but if they want to refund my money, go ahead... I wouldn't waste time having ten different accounts, it's impossible to monitor them all. Two good providers should provide all the redundency you need, and DIDs can't be redundent anyway. I think you will like Teliax. -- Chris Mason ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 18
I had deployed asterisk on my Toshiba Satellite laptop and it was running successfully, later I tried migrating to postgreSQL database, towards that i deployed the database server also on the same laptop and tried to configure the files as required. Somewhere i went wrong and now my asterisk implementation is not running giving this error message [cdr_pgsql.so]Dec 4 11:56:02 WARNING[3839]: loader.c:258 ast_load_resource: libpq.so.4: cannot open shared object file: No such file or directory Dec 4 11:56:02 WARNING[3839]: loader.c:440 load_modules: Loading module cdr_pgsql.so failed! I examined the earlier configured cdr_pgsql.conf file and everything seems to be in order there, I am a newbie in this asterisk arena so some help or guidance as to what might be possibly wrong would be appreciated. Regards hrishi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: missing libpq.so.4
hrishikesh shrivastaw wrote: [cdr_pgsql.so]Dec 4 11:56:02 WARNING[3839]: loader.c:258 ast_load_resource: libpq.so.4: cannot open shared object file: No such file or directory Dec 4 11:56:02 WARNING[3839]: loader.c:440 load_modules: Loading module cdr_pgsql.so failed! Check that you've installed libpq. That's the package that includes the shared library libpq.so.4, at least on my system. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
