Re: [asterisk-users] zap destroy
On Wed, Oct 01, 2008 at 01:39:29PM -0500, Jeff Peeler wrote: Nope, that's the best you can do without restarting Asterisk. Is requiring two restarts reproducible? I'd really like to see console output with verbosity and debug set to 4 on chan_dahdi, preferably while only using zap channels. Jeff is asking specifically about chan_dahdi and not about chan_zap because he fixed this issue and several related issues (e.g.: 'zap restart'/'dahdi restart' not working for a system with PRI, or after a configuration error). Though he fixed them at the specific version when chan_zap was renamed chan_dahdi and hence if you find those symptoms in chan_zap you probably still use the buggy version. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1)
On Wed, Oct 01, 2008 at 07:03:06PM -0400, Steve Totaro wrote: I own this combination of 1s and 0s. 111010010010101001001. Now please, 0x1B1254F81F , what's so novel about this? What is it supposed to do? Why is it not trivial? If you attempts to mock patents you won't get very far when you are faced with one that actually represents some original thinking (behind a thick layer of legalese). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1)
You need to lighten up buddy. The point is, who owns a series of ones and zeros??? Is your series of one's and zeros better than someone else' ones and zeros? Why because you have more, or the order is different? Take a vacation, get some tail or something. Geez. You are always riding posts, often with misinformation and never clear it up, or quote just the pieces of my post that help your argument and snip. I lose personal respect for people who behave this way. It's like CNN or FOX news. BTW, define trivial for me please. Is a race car engine trivial to a race car mechanic, of course. Is a race car engine trivial to my 8 year old niece, probably not so much. Let's try to use words that have more meaning and loosen up, seriously. Thanks, Steve Totaro On Thu, Oct 2, 2008 at 2:59 AM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Wed, Oct 01, 2008 at 07:03:06PM -0400, Steve Totaro wrote: I own this combination of 1s and 0s. 111010010010101001001. Now please, 0x1B1254F81F , what's so novel about this? What is it supposed to do? Why is it not trivial? If you attempts to mock patents you won't get very far when you are faced with one that actually represents some original thinking (behind a thick layer of legalese). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] B410p question
Hello list, I have got an asterisk box installed working ok with an b410p card to make and receive isdn calls. All works ok, but when a call is answer and the person starts to speak, always I can ear a beep during the call. This beep is ear some times in about 30 seconds between each beep. Pasted bellow I send /etc/misdn-init.conf and /etc/asterisk/misdn.conf Any clue will be apreciated. Thanks. VoipCrazy - My /etc/misdn-init.conf - # # Configuration file for your misdn hardware # # Usage: /usr/sbin/misdn-init start|stop|restart|config|scan|help # # # Card Settings # # Syntax: card=number,type[,option...] # #number count your cards beginning with 1 #type either 0x1,0x4 or 0x8 for your hfcmulti hardware, # or the name of your card driver module. #option ulaw - uLaw (instead of aLaw) # dtmf - enable DTMF detection on all B-channels # # pcm_slave - set PCM bus into slave mode #If you have a set of cards, all wired via PCM. Set #all cards into pcm_slave mode and leave one out. #The left card will automatically be Master. # # ignore_pcm_frameclock - this can be set in conjunction with # pcm_slave. If this card has a # PCI Bus Position before the Position # of the Master, then this port cannot # yet receive a frameclock, so it must # ignore the pcm frameclock. # # rxclock- use clocking for pcm from ST Port # crystalclock - use clocking for pcm from PLL (genrated on board) # watchdog - This dual E1 Board has a Watchdog for #transparent mode # # card=1,0x4 # # Port settings # # Syntax: port_type=port_number[,port_number...] # #port_typete_ptp - TE-Mode, PTP # te_ptmp - TE-Mode, PTMP # te_capi_ptp - TE-Mode (capi), PTP # te_capi_ptmp- TE-Mode (capi), PTMP # nt_ptp - NT-Mode, PTP # nt_ptmp - NT-Mode, PTMP #port_number port that should be considered # #te_ptmp=1,2,3,4 #te_ptmp=1,2 te_ptp=1,2,3,4 # # Port Options # # Syntax: option=port_number,option[,option...] # #option master_clock - use master clock for this S/T interface # (only once per chip, only for HFC 8/4) # optical - optical (only HFC-E1) # los - report LOS (only HFC-E1) # ais - report AIS (only HFC-E1) # slip - report SLIP (only HFC-E1) # nocrc4- turn off crc4 mode use double frame instead # (only HFC-E1) # # The master_clock option is essential for retrieving and transmitting # faxes to avoid failures during transmission. It tells the driver to # synchronize the Card with the given Port which should be a TE Port and # connected to the PSTN in general. # option=1,master_clock #option=2,ais,nocrc4 #option=3,optical,los,ais,slip # # General Options for your hfcmulti hardware # # poll=number # #Only one poll value must be given for all cards. #Give the number of samples for each fifo process. #By default 128 is used. Decrease to reduce delay, increase to #reduce cpu load. If unsure, don't mess with it!!! #Valid is 32, 64, 128, 256. # # dsp_poll=number # This is the poll option which is used by mISDN_dsp, this might # differ from the one given by poll= for the hfc based cards, since # they can only use multiples of 32, the dsp_poll is dependant on # the kernel timer setting which can be found in the CPU section # in the kernel config. Defaults are there either 100Hz, 250Hz # or 1000Hz. If your setting is either 1000 or 250 it is compatible # with the poll option for the hfc chips, if you have 100 it is # different and you need here a multiple of 80. # The default is to have no dsp_poll option, then the dsp itself # finds out which option is the best to use by itself # # pcm=number # #Give the id of the PCM bus. All PCM busses with the same ID #are expected to be connected and have equal slots. #Only one chip of the PCM bus must be master, the others slave. # # debug=number # #Enable debugging (see hfc_multi.h for debug options). # # dsp_options=number # # set this to 2 and you'll have software bridging instead of # hardware bridging. # # # dtmfthreshold=milliseconds # # Here you can tune the sensitivity of the dtmf tone recognizer. # # timer=1|0 # # set this to 1 if you want
Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1)
On Thu, Oct 02, 2008 at 04:23:26AM -0400, Steve Totaro wrote: You need to lighten up buddy. The point is, who owns a series of ones and zeros??? Is your series of one's and zeros better than someone else' ones and zeros? Why because you have more, or the order is different? You, for instance, might have copyright on the above paragraph you wrote here. The number you wrote there is not even long enough even for that: http://en.wikipedia.org/wiki/AACS_encryption_key_controversy Take a vacation, get some tail or something. Geez. You are always riding posts, often with misinformation and never clear it up, or quote just the pieces of my post that help your argument and snip. I lose personal respect for people who behave this way. It's like CNN or FOX news. BTW, define trivial for me please. Is a race car engine trivial to a race car mechanic, of course. Is a race car engine trivial to my 8 year old niece, probably not so much. A patent should be non-trivial to someone of the same trade. A patent for a race car should be something that is non-trivial for a race-car enggineer. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to find the CDR call start time value
Can anyone suggest how I can find the value of the call start time that will be logged by CDR in the dialplan? I've taken a look through the variables but I can't see anything that seems to hold this? The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Problem
Dear Sir,
I have the following Scenario:
1- I have a DID number from Voxbone mapped to my asterisk server with RFC
2833 protocol used for DTMF
2- On asterisk Server I configured an incoming peer that receives calls from
VoxBone and send calls to a2billing context as follow:
*sip.conf*
[sip_proxy1]
type=peer
context=a2billing
host=81.201.82.39
dtmfmode=RFC2833
rfc2833compensate=yes
*extensions.conf*
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,Playback(AR_GetGiveToID)
exten = _X.,22,Wait(2)
exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,24,Wait(2)
exten = _X.,25,Playback(/tmp/asterisk-recording)
exten = _X.,26,Wait(2)
exten = _X.,27,Hangup
My problem is that when entring the PIN number I did not notice that any
DTMF digits has been sent from VoxBone to my asterisk server, and the IVR
continue asking to enter the PIN number all the time as you can see in the
below log messages:
-
-- SIP/voxbone.com-0a02e0d8 Playing 'prepaid-enter-pin-number'
(language 'en')
a2billing.php: file:Class.A2Billing.php - line:1790 - RES DTMF :
a2billing.php: file:Class.A2Billing.php - line:1794 - CARDNUMBER ::
a2billing.php: file:Class.A2Billing.php - line:1798 -
PREPAID-NO-CARD-ENTERED
a2billing.php: file:Class.A2Billing.php - line:1780 -
PREPAID-NO-CARD-ENTERED
a2billing.php: file:Class.A2Billing.php - line:1788 - Requesting DTMF,
CARDNUMBER_LENGTH_MAX 15
-- SIP/voxbone.com-0a02e0d8 Playing 'prepaid-enter-pin-number'
(language 'en')
a2billing.php: file:Class.A2Billing.php - line:1790 - RES DTMF :
a2billing.php: file:Class.A2Billing.php - line:1794 - CARDNUMBER ::
a2billing.php: file:Class.A2Billing.php - line:1798 -
PREPAID-NO-CARD-ENTERED
a2billing.php: file:Class.A2Billing.php - line:1780 -
PREPAID-NO-CARD-ENTERED
a2billing.php: file:Class.A2Billing.php - line:1788 - Requesting DTMF,
CARDNUMBER_LENGTH_MAX 15
-- SIP/voxbone.com-0a02e0d8 Playing 'prepaid-enter-pin-number'
(language 'en')
What do you think the issue could be?
Regards
___
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Re: [asterisk-users] RTP sent before the INVITE ACK (for voicemail app)
On Wed, Oct 1, 2008 at 5:37 PM, tic tac [EMAIL PROTECTED] wrote: Thanks, in my case though it looks like the originating party (polycom softphone) is hearing a clipped voicemail prompt because of that; should I look into having that fixed into their firmware? As a workaround, I was thinking to just add a few seconds delay in app_voicemail, or wait through AGI before calling voicemail, makes sense? Yes. It's fairly standard practice to add a Wait(2) or Wait(3) at the start of a call Asterisk is generating direct audio on. This gives the RTP stream a chance to get sorted out. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find the CDR call start time value
HI Steven,
You can get call start time by ${CDR(start)} .
For more information of asterisk variables , please check out
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
Thanks,
Krunal Patel
On Thu, Oct 2, 2008 at 3:08 PM, Steve Hanselman [EMAIL PROTECTED]wrote:
Can anyone suggest how I can find the value of the call start time that
will be logged by CDR in the dialplan?
I've taken a look through the variables but I can't see anything that seems
to hold this?
The information contained in this email is intended for the personal and
confidential use
of the addressee only. It may also be privileged information. If you are
not the intended
recipient then you are hereby notified that you have received this document
in error and
that any review, distribution or copying of this document is strictly
prohibited. If you have
received this communication in error, please notify Brendata immediately
on:
+44 (0)1268 466100, or email '[EMAIL PROTECTED]'
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Registered Office as above. Registered in England No. 2764339
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Re: [asterisk-users] ATA for large networks
I personnlay found that marc is better than google when searching mailing lists : http://marc.info/?l=asterisk-usersr=1w=2 What is the best-recommended resource for searching archives of this mailing list? Thanks for your time ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find the CDR call start time value
That's exactly what I was looking for, I'd found this
http://www.voip-info.org/wiki/view/Asterisk+variables which seems to be
a partial copy of the same thing.
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Krunal
Patel
Sent: 02 October 2008 11:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to find the CDR call start time value
HI Steven,
You can get call start time by ${CDR(start)} .
For more information of asterisk variables , please check out
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
Thanks,
Krunal Patel
On Thu, Oct 2, 2008 at 3:08 PM, Steve Hanselman [EMAIL PROTECTED]
wrote:
Can anyone suggest how I can find the value of the call start time that
will be logged by CDR in the dialplan?
I've taken a look through the variables but I can't see anything that
seems to hold this?
The information contained in this email is intended for the personal and
confidential use
of the addressee only. It may also be privileged information. If you are
not the intended
recipient then you are hereby notified that you have received this
document in error and
that any review, distribution or copying of this document is strictly
prohibited. If you have
received this communication in error, please notify Brendata immediately
on:
+44 (0)1268 466100, or email '[EMAIL PROTECTED]'
Brendata (UK) Ltd
Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK
Registered Office as above. Registered in England No. 2764339
See our current vacancies at www.brendataco.uk
___
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The information contained in this email is intended for the personal and
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that any review, distribution or copying of this document is strictly
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+44 (0)1268 466100, or email '[EMAIL PROTECTED]'
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Registered Office as above. Registered in England No. 2764339
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Re: [asterisk-users] rebooting snoms in 1.6
With Asterisk 1.4 I could use commands like: /usr/sbin/asterisk -rx sip notify reboot-snom mjc_home to reboot a snom phone. Now, with 1.6, when I try that, I get: Unable to find notify type 'reboot-snom' Command 'sip notify reboot-snom mjc_home' failed. Do I need to add some magic to sip_notify.conf? I haven't quite figured out how to make it work. Found it. I needed: ; Untested - from Snom docs [reboot-snom] Event=reboot Content-Length=0 - Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ultramonkey LVS + asterisk
hi, has anyone implemented ultramonkey with asterisk? do i really need to setup fwmark as discussed in the url below? thanks! http://www.gossamer-threads.com/lists/lvs/users/20871 regards, ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IP address on mysql cdr
hi, is it possible to store the IP address of the caller in the CDR? how about the end date/time? thank you. regards, ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rebooting snoms in 1.6
Dr. Michael J. Chudobiak [EMAIL PROTECTED] writes: With Asterisk 1.4 I could use commands like: /usr/sbin/asterisk -rx sip notify reboot-snom mjc_home to reboot a snom phone. Now, with 1.6, when I try that, I get: Unable to find notify type 'reboot-snom' Command 'sip notify reboot-snom mjc_home' failed. Do I need to add some magic to sip_notify.conf? I haven't quite figured out how to make it work. I believe it works the same as in 1.4. You can add something like this to sip_notify.conf: [reboot-snom] Event=check-sync\;reboot=true Content-Length=0 [snom-check-cfg] Event=check-sync\;reboot=false Content-Length=0 /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF issue
Dear All, What could be the problem if I try to send DTMF in RFC2833 format to my asterisk server and it replies back with 603 error message? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Dropping SIP support?
Michael Graves schrieb: Earlier today I glanced at Junction Networks blog and was surprised to find a post indicating that Cisco was dropping SIP support in their 79xx series phones. Here's t link: http://www.junctionnetworks.com/blog/charlotte/2008/09/19/junction-netwo rks-lab-cisco-7960-phones Is this true? What are they thinking? Only SCCP? AFAIK the other way around is true. Cisco is dropping SCCP. The new firmware is for SIP only but it's with some Cisco extensions as the latest CCMs are using SIP as preferred protocol. Could be that Cisco drops the standard SIP FW though. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can Block a pri channel
Dwayne Hubbard wrote: Sean is correct I *never* get tired of hearing/reading that. -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can Block a pri channel
On Thu, Oct 2, 2008 at 9:40 AM, Sean Bright [EMAIL PROTECTED] wrote: Dwayne Hubbard wrote: Sean is correct I *never* get tired of hearing/reading that. -- Sean Bright [EMAIL PROTECTED] Insurance is a huge scam. They are betting for you by taking your money, and you are betting against yourself by paying them. -- Thanks, Steve Totaro 1 (888) 777-1888 (Toll Free) 1 (240) 938-1212 (Cell) 1 (202) 436-9784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Failover System
Redfone is not much good unless you have more than one Asterisk box. Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Wed, Oct 1, 2008 at 10:47 PM, Darren Sessions [EMAIL PROTECTED]wrote: I agree that an OpenSER solution on top of Asterisk for a 120 users is massive overkill to say the least. High calls-per-second? Multiple Asterisk servers? Multiple vendors? Advanced LCR? or plans for any of that in the near future? Then I think it makes sense to look at fronting Asterisk with OpenSER for such a small amount of users. Asterisk can do everything you'll need it to do otherwise. - D _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 1, 2008, at 7:44 PM, Alex Balashov wrote: Jai Rangi wrote: Openser? for 120 user? I would not do that. This would be an extra layer to configure, support, maintain and one more layer to debug if things go wrong. Its like spending a Dollar when you can be done with a quarter. (my 2 cents) All depends on how important those 120 users are. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel-1.4.1 error cross compile
I have CLFS ARM cross toolchain with uClibc and I have installed asterisk on it now I want to compile zaptel-1.4 I got this error clfs:/mnt/clfs/sources/zaptel-1.4.1$ make make[1]: Entering directory `/mnt/clfs/sources/zaptel-1.4.1/menuselect' checking build system type... i686-pc-linux-gnu checking host system type... i686-pc-linux-gnu checking for gcc... arm-unknown-linux-gnu-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... configure: error: cannot run C compiled programs. If you meant to cross compile, use `--host'. See `config.log' for more details. make[1]: *** [autoconfig.h] Error 1 make[1]: Leaving directory `/mnt/clfs/sources/zaptel-1.4.1/menuselect' make: *** [menuselect/menuselect] Error 2 clfs:/mnt/clfs/sources/zaptel-1.4.1$ Regards, Satish Patel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.1 error cross compile
On Thu, Oct 02, 2008 at 10:19:00AM -0400, satish patel wrote: I have CLFS ARM cross toolchain with uClibc and I have installed asterisk on it now I want to compile zaptel-1.4 I got this error clfs:/mnt/clfs/sources/zaptel-1.4.1$ make make[1]: Entering directory `/mnt/clfs/sources/zaptel-1.4.1/menuselect' checking build system type... i686-pc-linux-gnu checking host system type... i686-pc-linux-gnu checking for gcc... arm-unknown-linux-gnu-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... configure: error: cannot run C compiled programs. If you meant to cross compile, use `--host'. See `config.log' for more details. make[1]: *** [autoconfig.h] Error 1 make[1]: Leaving directory `/mnt/clfs/sources/zaptel-1.4.1/menuselect' make: *** [menuselect/menuselect] Error 2 clfs:/mnt/clfs/sources/zaptel-1.4.1$ Could you please try a newer version of zaptel 1.4? There have been many changes in the build system of zaptel 1.4 since 1.4.1 . It would also help if you give your configure command and/or environment. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is SIPPEER curcalls working for you ? (was: Ongoing calls with SIPPEER, curcalls)
Hello,
Has anyone successfully used this SIPPEER function ?
exten = _753X,n,Set(foo=${SIPPEER(${EXTEN}:curcalls)})
Then, did you get a meaningful value ?
I suspect my understanding of it is incorrect as I would say that if an
extension is on call with someone else, curcalls shall return 1 (which it
doesn't here as it returns 0).
Regards
2008/10/1 Doug Lytle [EMAIL PROTECTED]
Olivier wrote:
- curcalls is not set to what I was thinking (I misunderstood its
definition in voip-info.org http://voip-info.org, as I comply to
call-limit setting requirement)
- something else
My very basic testing, I'm not able to get a value either. I won't
have access to my testing system until later on this evening. I'll give
it another try then.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Is SIPPEER curcalls working for you ? (was: Ongoing calls with SIPPEER, curcalls)
Olivier wrote: Hello, Has anyone successfully used this SIPPEER function ? Olivier, I wasn't able to do any testing on it last night, I'll give it a try over this upcoming weekend and let you know what I find. 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 http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.1 error cross compile
Regards,
Satish Patel
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Thu, Oct 02, 2008 at 10:19:00AM -0400, satish patel wrote:
I have CLFS ARM cross toolchain with uClibc and I have installed asterisk on
it now I want to compile zaptel-1.4
I got this error
clfs:/mnt/clfs/sources/zaptel-1.4.1$ make
make[1]: Entering directory `/mnt/clfs/sources/zaptel-1.4.1/menuselect'
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... arm-unknown-linux-gnu-gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... configure: error: cannot run C
compiled programs.
If you meant to cross compile, use `--host'.
See `config.log' for more details.
make[1]: *** [autoconfig.h] Error 1
make[1]: Leaving directory `/mnt/clfs/sources/zaptel-1.4.1/menuselect'
make: *** [menuselect/menuselect] Error 2
clfs:/mnt/clfs/sources/zaptel-1.4.1$
Could you please try a newer version of zaptel 1.4? There have been many
changes in the build system of zaptel 1.4 since 1.4.1 .
It would also help if you give your configure command and/or
environment.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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clfs:/mnt/clfs/sources/zaptel-1.4.1$ ./configure --host=${CLFS_TARGET}
--prefix=/usr
configure: WARNING: If you wanted to set the --build type, don't use --host.
If a cross compiler is detected then cross compile mode will be used.
checking for arm-unknown-linux-gnu-gcc... arm-unknown-linux-gnu-gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... yes
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether arm-unknown-linux-gnu-gcc accepts -g... yes
checking for arm-unknown-linux-gnu-gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... arm-unknown-linux-gnu-gcc -E
checking for a BSD-compatible install... /usr/bin/install -c
checking whether ln -s works... yes
checking for GNU make... make
checking for grep... /bin/grep
checking for sh... /bin/sh
checking for ln... /bin/ln
checking for wget... /usr/bin/wget
checking for grep that handles long lines and -e... (cached) /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking for initscr in -lcurses... yes
checking curses.h usability... yes
checking curses.h presence... yes
checking for curses.h... yes
checking for initscr in -lncurses... yes
checking for curses.h... (cached) yes
checking for newtBell in -lnewt... no
checking for usb_init in -lusb... no
configure: creating ./config.status
config.status: creating build_tools/menuselect-deps
config.status: creating makeopts
configure: *** Zaptel build successfully configured ***
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Re: [asterisk-users] DTMF Problem
On Oct 2, 2008, at 5:27 AM, michel freiha wrote: Dear Sir, I have the following Scenario: 1- I have a DID number from Voxbone mapped to my asterisk server with RFC 2833 protocol used for DTMF 2- On asterisk Server I configured an incoming peer that receives calls from VoxBone and send calls to a2billing context as follow: sip.conf [sip_proxy1] type=peer context=a2billing host=81.201.82.39 dtmfmode=RFC2833 rfc2833compensate=yes Try adding: relaxdtmf=yes to the peer Fred Posner [EMAIL PROTECTED] Tel: +1 (212) 937-7844 x501 www.teamforrest.com Using VoIP? SIP:[EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.1 error cross compile
On Thu, Oct 02, 2008 at 10:51:37AM -0400, Satish Patel wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
As I wrote:
Could you please try a newer version of zaptel 1.4? There have been many
changes in the build system of zaptel 1.4 since 1.4.1 .
But in your reply:
clfs:/mnt/clfs/sources/zaptel-1.4.1$ ./configure --host=${CLFS_TARGET}
--prefix=/usr
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Asterisk custom functions
On Wednesday 01 October 2008 23:58:41 Max Alex wrote: Hi All, i have centos5 system, i have installed asterisk 1.4 branch. i havedone realtime connection with odbc to pgsql. i have created custom functions in func_odbc.conf, all dsn setup and connection is working fine, but custom functions are not being registered to asterisk. i have given queries to functions and using that functions in dialplan. but it is always gives me function is not registered. can any body explain how to register custom functions in asterisk? module reload func_odbc.so -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.1 error cross compile
Regards,
Satish Patel
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Thu, Oct 02, 2008 at 10:51:37AM -0400, Satish Patel wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
As I wrote:
Could you please try a newer version of zaptel 1.4? There have been many
changes in the build system of zaptel 1.4 since 1.4.1 .
But in your reply:
clfs:/mnt/clfs/sources/zaptel-1.4.1$ ./configure
--host=${CLFS_TARGET} --prefix=/usr
I wanted to show you what option i used now i have download zaptel-1.4.12.1
clfs:/mnt/clfs/sources/zaptel-1.4.12.1$ ./configure
--host=${CLFS_TARGET} --prefix=/usr
configure: WARNING: If you wanted to set the --build type, don't use --host.
If a cross compiler is detected then cross compile mode will be used.
checking for arm-unknown-linux-gnu-gcc... arm-unknown-linux-gnu-gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... yes
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether arm-unknown-linux-gnu-gcc accepts -g... yes
checking for arm-unknown-linux-gnu-gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... arm-unknown-linux-gnu-gcc -E
checking for a BSD-compatible install... /usr/bin/install -c
checking whether ln -s works... yes
checking for GNU make... make
checking for grep... /bin/grep
checking for sh... /bin/sh
checking for ln... /bin/ln
checking for wget... /usr/bin/wget
checking for grep that handles long lines and -e... (cached) /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking for initscr in -lcurses... yes
checking curses.h usability... yes
checking curses.h presence... yes
checking for curses.h... yes
checking for initscr in -lncurses... yes
checking for curses.h... (cached) yes
checking for newtBell in -lnewt... no
checking for usb_init in -lusb... no
configure: creating ./config.status
config.status: creating build_tools/menuselect-deps
config.status: creating makeopts
config.status: creating build_tools/make_firmware_object
configure: *** Zaptel build successfully configured ***
clfs:/mnt/clfs/sources/zaptel-1.4.12.1$ export
clfs:/mnt/clfs/sources/zaptel-1.4.12.1$ make ARCH=arm KVERS=2.6.22.6
CROSS_COMPILE=${CLFS_TARGET}- modules zaptel
clfs:/mnt/clfs/sources/zaptel-1.4.12.1$ export KVERS=2.6.22.6
clfs:/mnt/clfs/sources/zaptel-1.4.12.1$ make
make: Warning: File `Makefile' has modification time 6.1e+05 s in the future
make[1]: Entering directory `/mnt/clfs/sources/zaptel-1.4.12.1/menuselect'
checking build system type... i686-pc-linux-gnu
checking host system type... i686-pc-linux-gnu
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables...
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ANSI C... none needed
checking for GNU make... make
checking for asprintf... yes
checking for getloadavg... yes
checking for setenv... yes
checking for strcasestr... yes
checking for strndup... yes
checking for strnlen... yes
checking for strsep... yes
checking for strtoq... yes
checking for unsetenv... yes
checking for vasprintf... yes
checking how to run the C preprocessor... gcc -E
checking for egrep... grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking for initscr in -lcurses... yes
checking curses.h usability... yes
checking curses.h presence... yes
checking for curses.h... yes
checking for initscr in -lncurses... yes
checking for curses.h... (cached) yes
checking for pkg-config... pkg-config
Package gtk+-2.0 was not found in the pkg-config search path.
Perhaps you should add the directory containing `gtk+-2.0.pc'
to the PKG_CONFIG_PATH environment variable
No package 'gtk+-2.0' found
configure: creating ./config.status
config.status: creating makeopts
config.status: creating autoconfig.h
configure: configuring in mxml
configure: running /bin/sh './configure' --prefix=/usr/local 'CC='
'LD=' 'AR=' 'CFLAGS=' --cache-file=/dev/null --srcdir=.
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are
Re: [asterisk-users] zap destroy
- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Oct 01, 2008 at 01:39:29PM -0500, Jeff Peeler wrote: Nope, that's the best you can do without restarting Asterisk. Is requiring two restarts reproducible? I'd really like to see console output with verbosity and debug set to 4 on chan_dahdi, preferably while only using zap channels. Jeff is asking specifically about chan_dahdi and not about chan_zap because he fixed this issue and several related issues (e.g.: 'zap restart'/'dahdi restart' not working for a system with PRI, or after a configuration error). Though he fixed them at the specific version when chan_zap was renamed chan_dahdi and hence if you find those symptoms in chan_zap you probably still use the buggy version. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Yes, the new changes will be in 1.4.22. I continually have to remind myself that users aren't running the most up to date code. Jeff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zap destroy
On Oct 2, 2008, at 9:10 AM, Jeff Peeler wrote: - Tzafrir Cohen [EMAIL PROTECTED] wrote: Yes, the new changes will be in 1.4.22. I continually have to remind myself that users aren't running the most up to date code. Once 1.4.22 comes out I will report if I am still having those issues. Daniel Jeff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - Is sip.instance useful ?
Hi, I've seen some hardphones or Softswitchs now support this sip.instance feature : http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt I don't really see any convincing use of this draft but I would be curious to share thoughts on it. Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queue question
When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy Fewest Calls working for a couple of mouths, and a new agent has been added this week in the queue and he is receiving all the incomings calls. How could I solve that? Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue question
On Thu, Oct 2, 2008 at 7:32 PM, voip crazy [EMAIL PROTECTED] wrote: When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy Fewest Calls working for a couple of mouths, and a new agent has been added this week in the queue and he is receiving all the incomings calls. How could I solve that? I do a nightly restart, however i suppose that module reload app_queue.so would do the trick :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Problem
Hi; This problem I suffered from it for long time, it needs some work from ur side to resolve it, I will give u all the factors that will help u to fix it, and u need to work on it one after one in care: 1) Disable x-windows, gnome, ... at least for all testing. This is very important to be done. 2) Disable all devices not used, USB, Video, etc. (If possible to be done), but at least you should disable the x-windows. 3)Give lower IRQ for the digium card (for higher priority). 4) echocancelwhenbridged=no in zapata.conf, bcz this can cause problems when not needed, only to be used as a last resort for echo issues. 5) Raise gain in hardware, return to 0 in software. In hardware, the file (/etc/sysconfig/zaptel and /etc/modprobe.conf). 6) Run fxo tune and lower the gain in software, this will remove the static sound on the line (static noise). fxotune (type man fxotune to read about it). fxotune -i -vv -b 3 -i is the configuration mode -vv for vesibility -b for testing at module 3 Note: asterisk should be stopped before running the fxotune. 6) set opermode=KUWAIT (ir country) in /etc/sysconfig/zaptel and /etc/modprobe.conf If I am in ur case, I will disable x-windows and then I will make gain = 0 in the software and increase it only in the hardware. Also, I will set the opermode=my country. Do not forget to stop asterisk and run the fxotune to remove the statc noise. Looking to hear from you if that problem resolved. Regards Bilal Dear Sir, I have the following Scenario: 1- I have a DID number from Voxbone mapped to my asterisk server with RFC 2833 protocol used for DTMF 2- On asterisk Server I configured an incoming peer that receives calls from VoxBone and send calls to a2billing context as follow: sip.conf [sip_proxy1] type=peer context=a2billing host=81.201.82.39 dtmfmode=RFC2833 rfc2833compensate=yes Try adding: relaxdtmf=yes to the peer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel-1.4.1 error cross compile
Regards,
Satish Patel
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Thu, Oct 02, 2008 at 11:33:01AM -0400, Satish Patel wrote:
Regards,
Satish Patel
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
On Thu, Oct 02, 2008 at 10:51:37AM -0400, Satish Patel wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
As I wrote:
Could you please try a newer version of zaptel 1.4? There have been many
changes in the build system of zaptel 1.4 since 1.4.1 .
But in your reply:
clfs:/mnt/clfs/sources/zaptel-1.4.1$ ./configure
--host=${CLFS_TARGET} --prefix=/usr
I wanted to show you what option i used now i have download zaptel-1.4.12.1
clfs:/mnt/clfs/sources/zaptel-1.4.12.1$ ./configure
--host=${CLFS_TARGET} --prefix=/usr
configure: WARNING: If you wanted to set the --build type, don't use --host.
If a cross compiler is detected then cross compile mode will be used.
I don't know much about cross-compiling, but this warning scares me.
I have a feeling you're doing something wrong.
Anyway, if you want to avoid the whole menuselect mess, take a look at
http://bugs.digium.com/13132
Remove the subdirectory menuselect and put the makefile and script from
that bug report there instead. Run:
make -C menuselect dummies
Then it should behave just like the original. At least theoretically.
You may need to instruct it to take data from other XML files. See the
calls to the function parse_menuselect_xml_file() in the end.
Let me know if it worked ;-)
I'll see if someone else will pick it up on-list as both cross-compiling
and menuselect are not my preffered code.
for experiment i have download 1.2.27 current version of zaptel
./configure --host=${CLFS_TARGET} --prefix=/usr
make ARCH=arm CROSS_COMPILE=${CLFS_TARGET}-
make ARCH=arm CROSS_COMPILE=${CLFS_TARGET}- DESTDIR=${CLFS} install
it has installed module
clfs:/mnt/clfs/sources/zaptel-1.2.27$ ls -l ../../lib/modules/2.6.22.6/misc/
total 476
-rw-r--r-- 1 clfs clfs 67566 Sep 1 18:46 pciradio.ko
-rw-r--r-- 1 clfs clfs 92753 Sep 1 18:46 tor2.ko
-rw-r--r-- 1 clfs clfs 19267 Sep 1 18:46 torisa.ko
-rw-r--r-- 1 clfs clfs 15542 Sep 1 18:46 wcfxo.ko
-rw-r--r-- 1 clfs clfs 18524 Sep 1 18:46 wct1xxp.ko
drwxr-xr-x 2 clfs clfs 4096 Sep 1 18:42 wct4xxp
drwxr-xr-x 2 clfs clfs 4096 Sep 1 18:42 wctc4xxp
-rw-r--r-- 1 clfs clfs 46475 Sep 1 18:46 wctdm.ko
drwxr-xr-x 2 clfs clfs 4096 Sep 1 18:42 wctdm24xxp
-rw-r--r-- 1 clfs clfs 40601 Sep 1 18:46 wcte11xp.ko
drwxr-xr-x 2 clfs clfs 4096 Sep 1 18:42 wcte12xp
-rw-r--r-- 1 clfs clfs 18531 Sep 1 18:46 wcusb.ko
-rw-r--r-- 1 clfs clfs 71372 Sep 1 18:46 zaptel.ko
-rw-r--r-- 1 clfs clfs 8250 Sep 1 18:46 ztd-eth.ko
-rw-r--r-- 1 clfs clfs 4883 Sep 1 18:46 ztd-loc.ko
-rw-r--r-- 1 clfs clfs 3204 Sep 1 18:46 ztdummy.ko
-rw-r--r-- 1 clfs clfs 13059 Sep 1 18:46 ztdynamic.ko
-rw-r--r-- 1 clfs clfs 10780 Sep 1 18:46 zttranscode.ko
but when i have build-root and run is root-image on ARM hardware and
try to install module i got error
root#insmod zaptel
insmod: cannot insert '/lib/modules/2.6.22.6/misc/zaptel.ko' : Invalid
module formate (-1): Exec format error
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[asterisk-users] dahdi-linux 2.0.0 and dahdi-tools 2.0.0 released
The Asterisk development team is pleased to announce the first offical release of the Digium Asterisk Hardware Device Interface (DAHDI). The list of packages released today includes: dahdi-linux 2.0.0 dahdi-tools 2.0.0 dahdi-linux-complete 2.0.0+2.0.0 Both dahdi-linux and dahdi-tools are required to enable DAHDI support in your system. You will need to install dahdi-linux first, then dahdi-tools, and finally you can configure and make Asterisk. dahdi-linux-complete is both dahdi-linux and dahdi-tools combined into one download as a convenience. You still need libpri for PRI support with Asterisk if you are using DAHDI. DAHDI is supported by Asterisk 1.4.22 and Asterisk 1.6.0. More detailed information about each of the packages is below. == dadhi-linux-2.0.0 == This is the first release of the DAHDI Linux kernel modules package, which replaces the kernel modules from Zaptel. The primary purpose of this release is to rename the package from Zaptel but in addition to several bug fixes some of the new features are: * Echo cancelers can now be applied per channel and selected at configuration time. * Channel memory allocation changed from one large block into smaller blocks in order to reduce out of memory errors on a system that has been running for some time. * Layout changes to support binary packaging. * Neon MWI support added to the wctdm24xxp driver. * Dropped support for Linux Kernel 2.4 as well as the torisa and wcusb drivers. For information on upgrading from Zaptel to this release, please see: http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt Known Issues * Reference counting is not currently done on echo canceler modules, and therefore it is possible for an administrator to unload an echo canceler module that is in use which could result in a crash. It is recommended to use /etc/init.d/dahdi start|stop to load and unload your drivers to eliminate exposure to this issue. http://bugs.digium.com/view.php?id=13504 * Cannot compile with CONFIG_DAHDI_NET or use DAHDI for data connections. http://bugs.digium.com/view.php?id=13542 === dahdi-tools-2.0.0 == This is the first release of the DAHDI userspace tools package, which replaces the userspace components of Zaptel. The primary purpose is to rename components from Zaptel to DAHDI and support binary packaging. The names and layouts of the configuration files have also changed. Please see UPGRADE.txt for more information. http://svn.digium.com/svn/dahdi/tools/tags/2.0.0/UPGRADE.txt === dahdi-linux-complete-2.0.0+2.0.0 === This release combines dahdi-linux and dahdi-tools into a single download, one-package installation process. Users who are installing DAHDI for the first time don't have to download and install the dahdi-linux and dahdi-tools packages separately. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.22 and 1.6.0 Released
The Asterisk.org development team is proud to announce the releases of Asterisk 1.4.22 and 1.6.0. = === Asterisk 1.4.22 = = Asterisk 1.4.22 includes a large number of bug fixes for the 1.4 release series of Asterisk. 1.4.22 also includes support for DAHDI. For more information about the transition from Zaptel to DAHDI, please see the following help file: http://svn.digium.com/view/asterisk/tags/1.4.22/Zaptel-to-DAHDI.txt?view=markup For a full listing of changes in this release, see the ChangeLog: http://svn.digium.com/view/asterisk/tags/1.4.22/ChangeLog?view=markup Asterisk 1.4.22 is available for immediate download from the Digium downloads site: http://downloads.digium.com/pub/telephony/asterisk/asterisk-1.4.22.tar.gz = = = === Asterisk 1.6.0 == = Asterisk 1.6.0 is the first official release of Asterisk 1.6. - --- Upgrade Information - - Asterisk 1.6 no longer supports Zaptel. It only contains support for DAHDI. For more information on this transition, please see the following help file: http://svn.digium.com/view/asterisk/tags/1.6.0/Zaptel-to-DAHDI.txt?view=markup There are a number of other important changes to be aware of when upgrading to Asterisk 1.6.0 from previous versions of Asterisk. For a listing of those things, please see UPGRADE.txt: http://svn.digium.com/view/asterisk/tags/1.6.0/UPGRADE.txt?view=markup - --- New Features - Asterisk 1.6.0 contains new features that were not previously available in an Asterisk release. For a full listing of the features that are included in Asterisk 1.6.0, please see the CHANGES file: http://svn.digium.com/view/asterisk/tags/1.6.0/CHANGES?view=markup A verbose listing of each individual change that was made in the development of Asterisk 1.6.0 is also available: http://svn.digium.com/view/asterisk/tags/1.6.0/ChangeLog?view=markup - --- Release Management -- - The Asterisk.org development team has decided on a new release management style for Asterisk 1.6. Previously, a release series was strictly feature frozen for its entire lifetime. The release management guidelines for Asterisk 1.6 were inspired by the Linux Kernel, among a number of other projects. Asterisk 1.6 is not feature frozen. Features will be added in point releases. However, effort will be made to ensure that the number of large changes is minimized in a single point release. Even as 1.6.0 is being released, the Asterisk development team is already working on what new things will be in 1.6.1 and beyond. To take a look ahead to see what features have been added for releases that have not yet been made, take a look at the trunk version of the CHANGES file: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=markup Even though new features will be added in point releases of Asterisk 1.6, that does not mean that any deprecated functionality will be removed as has been done between major releases in the past. In fact, we have decided that maintaining backwards compatibility is of the utmost importance for configuration and external interfaces. C API and ABI compatibility is not guaranteed between point releases. However, things like dialplan applications, functions, and AGI commands will not disappear just because there is a new and better way to accomplish the same thing. With Asterisk 1.4, once Asterisk 1.4.N is released, Asterisk 1.4.X is no longer supported, where X N. With Asterisk 1.6, the development team plans to maintain a total of 3 releases at a time. For example, the development team will support Asterisk 1.6.0, 1.6.1, and 1.6.2 until 1.6.3 is released. This means that for the time that 1.6.0 is supported, there may be 1.6.0.1, 1.6.0.2, etc. releases that include fixes for regressions found in 1.6.0. With Asterisk 1.4, the goal has been to make releases every 4 to 6 weeks. With Asterisk 1.6, we aim to release updates in a similar time frame, but it is likely that it will be closer to 6 to 8 weeks for Asterisk 1.6 due to a more strict beta and release candidate testing
[asterisk-users] Channels crossing...
I have a customer that is reporting that sometimes when they dial an outside line they can hear other conversations. At this moment I am assuming it only happens when they dial an outside number and not between extensions. They are using Asterisk 1.4.11, Zaptel 1.2.12.1 (just updated today) and Rhino-2.2.6 (Rhino 8 port card) on a CentOS 5 machine. Even with the update today it happened again. Is this a problem with Asterisk or Zaptel? How can I fix it? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels crossing...
Carlos Chavez wrote: Is this a problem with Asterisk or Zaptel? How can I fix it? Actually, sounds like an analog line with a bad punch down or frayed shielding and your getting cross talk. It may also be an issue at your provider. 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 http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOIP Provider
Hi All, Can anyone recommend a good VOIP provider in the Milwaukee/Chicago area? We need flat rate billing per line/trunk, trunking, did’s, and iax or G.729 compatibility. Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.7.5/1703 - Release Date: 10/2/2008 7:46 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP Provider
2008/10/2 Gregory Malsack [EMAIL PROTECTED] Hi All, Can anyone recommend a good VOIP provider in the Milwaukee/Chicago area? We need flat rate billing per line/trunk, trunking, did's, and iax or G.729 compatibility. Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.7.5/1703 - Release Date: 10/2/2008 7:46 AM Bandwidth.com is good, has flat rate, trunk, not sure about their stock of DIDs. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF
How can I know for sure if SIP Trunk Provider is sending DTMF 'inband' or 'rfc2833'? And more importantly if they could be sending both? If I specify 'inband' should they honor that? Thanks, Bart___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue question
Yes it is, every counter is set to zero: asterisk -rx module reload app_queue.so Regards, Daniel Arohuanca t.+51 1 994149553 Peru On Thu, Oct 2, 2008 at 12:05 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Thu, Oct 2, 2008 at 7:32 PM, voip crazy [EMAIL PROTECTED] wrote: When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy Fewest Calls working for a couple of mouths, and a new agent has been added this week in the queue and he is receiving all the incomings calls. How could I solve that? I do a nightly restart, however i suppose that module reload app_queue.so would do the trick :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP Provider
Hi. We recommend Fonet Global, they work with Asterisk many years ago and provide sip termination, DIDs, etc. At 03:39 p.m. 02/10/2008, Steve Totaro wrote: 2008/10/2 Gregory Malsack mailto:[EMAIL PROTECTED][EMAIL PROTECTED] Hi All, Can anyone recommend a good VOIP provider in the Milwaukee/Chicago area? We need flat rate billing per line/trunk, trunking, did's, and iax or G.729 compatibility. Thanks, Greg No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.7.5/1703 - Release Date: 10/2/2008 7:46 AM Bandwidth.com is good, has flat rate, trunk, not sure about their stock of DIDs. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels crossing...
I have seen and heard the recordings to prove crossed calls between people and agents in a busy call center, it was kind of funny with two agents trying to figure out what was going on and a very confused customer. I certainly would not rule out Asterisk, but you always start with the cables. On Thu, Oct 2, 2008 at 3:57 PM, Doug Lytle [EMAIL PROTECTED] wrote: Carlos Chavez wrote: Is this a problem with Asterisk or Zaptel? How can I fix it? Actually, sounds like an analog line with a bad punch down or frayed shielding and your getting cross talk. It may also be an issue at your provider. 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 http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi service start
I just downloaded the dahdi release and installed it. I removed anything zaptel I could find. /sbin , /lib/modules and others... when doing a service dahdi start it is still looking to ztcfg Why? I have look all about and cant determine why? service dahdi start Loading DAHDI hardware modules: wct4xxp: sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wct4xxp [FAILED] wcte12xp: sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wcte12xp [FAILED] wct1xxp: sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wct1xxp [FAILED] wcte11xp: sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wcte11xp [FAILED] wctdm24xxp: sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wctdm24xxp [FAILED] wcfxo: sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wcfxo [FAILED] wctdm: sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wctdm [FAILED] xpp_usb: [ OK ] No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: [ OK ] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Dropping SIP support?
They are probably referring to the fact that the base 7960 is End of Life and the 7960G is probably going to be EOL soon as well, so they won't offer new firmware at the EOL milestone. They have been replaced by the 7961. Completely different firmware and configuration, but there still is support for SIP. Stefan Gofferje wrote: Michael Graves schrieb: Earlier today I glanced at Junction Networks blog and was surprised to find a post indicating that Cisco was dropping SIP support in their 79xx series phones. Here's t link: http://www.junctionnetworks.com/blog/charlotte/2008/09/19/junction-netwo rks-lab-cisco-7960-phones Is this true? What are they thinking? Only SCCP? AFAIK the other way around is true. Cisco is dropping SCCP. The new firmware is for SIP only but it's with some Cisco extensions as the latest CCMs are using SIP as preferred protocol. Could be that Cisco drops the standard SIP FW though. Terve, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] uninstalling zaptel
What is the correct way to uninstall zaptel in the zaptel directory I can do make uninstall-modules which does just that but what about all the other files??? /etc/udev/rules/XX /etc/init.d/XX /sbin/ztXX and others doing a make uninstall gives an error. Is there anything that removes all those other files. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels crossing...
Hey Steve, it's been a while... On Thu, Oct 2, 2008 at 8:33 PM, Steve Totaro [EMAIL PROTECTED] wrote: I have seen and heard the recordings to prove crossed calls between people and agents in a busy call center, it was kind of funny with two agents trying to figure out what was going on and a very confused customer. I certainly would not rule out Asterisk, but you always start with the cables. I have just seen this myself on a pure voip and busy callcenter... with recordings to prove it too. I would not rule Asterisk either, as I do not have many cables to check! -- Nicolás Gudiño Buenos Aires - Argentina ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi service start
Jerry Geis wrote: wct4xxp: sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wct4xxp [FAILED] Hmm.. Something in /etc/modprobe.conf, /etc/modules.conf, or /etc/modprobe.d/? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] t1 cards
I presently need to connect a few channels of voice and data between multiple locations where I own the copper between them. Each location exceeds 300M from any other location. I'm thinking of generating T1's and running those between locations. If I use PC based cards wired back to back (I can do that, right?) what kind of distance can I expect to be able to span without needing repeaters? What inexpensive cards can you recommend for use with asterisk? I'm considering either digium or sangoma. Would I get any better performance if I used a sync-serial card connected to a separate csu/dsu? Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
