[Asterisk-Users] Transfer call via AMI or dialplan
At the moment when one of our users wants to transfer a call, they press the transfer button on the phone, enter the extension and away they go. Is there any way to do this via the AMI or dialplan ? I want them to push a button on the screen rather than using the phone itself. Julian ___ --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 voicemail problem with isdn avm fritz!card
Hello everyone,
I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz,
256MB) with a TDM40b a TDM04b and an avm fritz!card pci.
I experience a problem with voicemail: my messages are good unless the
incoming call comes from isdn, which means via the avm fritz!card. In
this case (and in this case only) the message is disjointed and I can
hear at most 1 second out of a 1 minute message.
If the message comes from TDM400 then the message is perfect (even
though I still have a problem to detect the end of the call but that's
no big deal)
If the incoming call is answered (and not sent to voicemail because busy
or unavail) the sound is perfect.
I hope you'll be able to help me.
Thanks
Benjamin SEBBAH
ADUNEO France
Here are my config files:
/etc/asterisk/capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=fr ;set default language
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
controller=1 ;capi controller number to use
group=9 ;dialout group
softdtmf=on ;enable/disable software dtmf detection, recommended
for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
echocancelold=yes;use facility selector 6 instead of correct 8
(necessary for older eicon drivers)
echotail=64 ;echo cancel tail setting
devices=2;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
and the interesting lines from /etc/asterisk/extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
PIERRE=Zap/1
MARC=SIP/marc
PATRICK=Zap/3
PROSPECT=Zap/2
OPENSPACE=Zap/4
FT_FREE=Zap/5
FT_ALICE=Zap/6
VOIP_FREE=Zap/7
VOIP_ALICE=Zap/8
NUMERIS=CAPI/ISDN1
[macro-repondeur]
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},15,rWw) ; Ring the interface,
15 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
;exten = s-NOANSWER,2,Goto(default,s,1); If they press #,
return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
;exten = s-BUSY,2,Goto(default,s,1); If they press #, return to
start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else
as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user
into VoicemailMain
[capi-in]
;standard: fait tout sonner
exten = 3090,1,Answer;
;exten = 3090,2,Macro(repondeur,8427,${OPENSPACE}${MARC}${PIERRE});
exten = 3090,2,Macro(repondeur,8427,${OPENSPACE}${PIERRE});
;Service technique
exten = 3091,1,Answer;
;exten = 3091,2,Macro(repondeur,3091,${OPENSPACE}${MARC});
exten = 3091,2,Macro(repondeur,3091,${OPENSPACE});
;Service commercial
exten = 3092,1,Answer;
exten = 3092,2,Macro(repondeur,3092,${PATRICK});
;Direction technique
exten = 3093,1,Answer;
;exten = 3093,2,Macro(repondeur,3093,${MARC});
exten = 3093,2,Macro(repondeur,3093,${OPENSPACE});
;non assigne pour le moment fait sonner uniquement le DECT
exten = 3094,1,Answer;
exten = 3094,2,Macro(repondeur,3094,${OPENSPACE});
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Re: [Asterisk-Users] Creating Queues on Asterisk server - Sending ingress calls off-net to either PSTN or another VoIP application - thoughts?
Hello Christopher, an Asterisk callback agent can be anywhere, even on a POTS number. He will have to register with a number that can reach him as far as Asterisk is concerned. I don't see the scenario you are proposing as particularly difficult to implement in Asterisk. Hope this helps l. On Mon, 19 Jun 2006 02:19:08 +0200, Christopher Aloi [EMAIL PROTECTED] wrote: Hello, Long time subscriber/reader of this list - thank you for all the great ideas. Scenario: We currently provide a hosted ACD system using Mitel phones (speaking the Minet protocol) to an NCI based server solution. The logic behind this choice was the emulation of key system features etc... Many of our clients have asked for basic call queue functionality: - Agents having the ability to login to a specific queue - Call distributed to that queue based on criteria - Basic reporting (ASA, AHT etc..) Solutions: - Flip the Mitel phones to load a SIP firmware and speak to AST (althought i'd love it, the powers that be probably won't) - Use the Asterisk queueing ability to send calls off network (AST) to the NCI platform (the Asterisk box can send these calls via SIP or TDM through a gateway). Goals: I'd like to create an Asterisk server running multiple queues for multiple tenants (or customers) that can provide the ability for agents to login remotely (either via an ingress call to AST or a www gui). The call flow would be similar to this: Agent#1 - logs into Mitel phone Agent#1 - Dials XXX XXX into Astersik Agent#1 - Hears a prompt on Asterisk to login to a specifc queue Agent#1 - Passes DTMF and becomes 'available' in the eyes of Asterisk Agent#1 - Is now in queue *repeat for three agents* Now, all three agents are in an available state to Asterisk, and logged into our one queue. If Asterisk receives a call on a specific DID it will attempt to send the goal to agent#1, if agent#1 rings three times or returns a 'busy here' the call will pass to agent#2 etc. The challenge I see will be configuring an off-network queue, is anyone working with a similar setup? Does anyone have any thoughts on how to better accomplish my goals? Thanks in advance. /Chris -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] GXP-2000 1.1.0.13 Issues
Grandstream have acknowledged that there is a problem with 1.1.0.13 on later phones (MAC's 00:0B:82:09:xx:xx I assume) and have advised me to wait for the next firmware release. So anyone with later phones (MAC's 00:0B:82:09:xx:xx), do not upgrade to 1.1.0.13. On Wed, 14 Jun 2006 [EMAIL PROTECTED] wrote: I have had 2 GXP-2000 for a while now and been slowly following the firmware releases made by Grandstream and am now up to 1.1.0.13. This version works really well on these 2 original phones (MAC's 00:0B:82:06:xx:xx), so I went ahead and ordered another 2 phones (MAC's 00:0B:82:09:xx:xx). One of these I upgraded to 1.1.0.13 (it came with 1.1.0.5) and pressed it into use. The Speaker phone does not work at all (no sound from the Speaker) and the phone completely hangs doing a soft-reboot, other than that the phone seems to work well. Unfortunatly I do not have a copy of 1.1.0.5 so cannot downgrade the phone. Has anyone else noticed these problems, or does anyone have a copy of 1.1.0.5. -Drew- ___ --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] show queue ... Invalid
Hi! I've added member to a queue like this, from queues.conf: member = SIP/[EMAIL PROTECTED] It works OK. But, after restaring I see in show queue that Members: SIP/[EMAIL PROTECTED] (Invalid) ... What does it mean? Why is it Invalid? BTW, reload command fixes it, so the member receives queue calls. Thanks! PS. 1.2.9.1 -- DSS5-RIPE DSS-RIPN 2:550/[EMAIL PROTECTED] 2:550/[EMAIL PROTECTED] xmpp:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://neva.vlink.ru/~dsh/ ___ --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] asttapi 0.10
Hi,
I have been playing around with the latest release of asttapi and I have
found the 'hangup' problem already reported to the list here
http://lists.digium.com/pipermail/asterisk-users/2006-May/151260.html
Apparently hangup should be done by making use of UserEvent commands. So I
have configured this context for being used when making calls from outlook:
[outlook]
exten = _X.,1,UserEvent(TAPI|TAPIEVENT: [~${UNIQUEID}] LINE_CALLSTATE
LINECALLSTATE_CONNECTED)
exten = _X.,2,Dial(SIP/[EMAIL PROTECTED] mailto:SIP/[EMAIL PROTECTED])
exten = _X.,3,UserEvent(TAPI|TAPIEVENT:[~${UNIQUEID}] LINE_CALLSTATE
LINECALLSTATE_HANGUP)
exten = _X.,4,Hangup
Dialling is ok, but outlook keeps on getting stuck in status
'dialling..'.Despite of Asterisk manager reporting the UserEvents, Asttapi
doesn't seem to be getting any information.
Now my question is..
Is it possible to hangup the outlook thing?
And if it is,
Why it is not working for me? Is it because the given configuration is
wrong? Is it because I'm using windows 2000 or outlook 2000 and I should try
a different version? (I have tried different versions of Asterisk with the
same result).
As an alternative I wouldn't mind at all to forget about UserEvents and just
close the outlook window after sending the call to Asterisk, if that's
possible.
Thanks a lot,
Victor.
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Re: [Asterisk-Users] Echo Problem with T411P
More than 128ms? 128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but definitely more than 16ms. No, 128ms = 1024 taps Like what sangoma offers. Ding, Ding, Ding, Ding! Okay, to be complete in my answers: No I do not get more than 128ms delay caused by European routing (I only threw that in as an example anyway), but asterisk's software cancellers only cancel 16ms, any more than that seems fairly buggy, and eats CPU. On the other hand, if that is a satellite link on span 3, you could easily get latency in excess of 1 second, which it should be the provider's responsibility to cancel, not the end user's IMHO. I also agree that the sangoma EC is excellent :) Do we know what E1/T1 hardware is in use here, and whether hardware EC is available? Steve ___ --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] Canreinvite
I will try your suggestion and I will let you know. Thank you On 6/18/06, Philippe Lindheimer [EMAIL PROTECTED] wrote:How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd on the specific channel to see where the rtp streams are going. Or ... if this is the only active channel on the box, just do a rtp debug. If the rtp stream is going through asterisk, it will be very obvious. If not, you won't see a constant flow of rtp debug messages going on. pFrom: Il Neofita [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30) exten = _40002,1,Dial(SIP/40002,30) From: Il Neofita [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 18 Jun 2006 05:22:35 -0400Subject: Re: [Asterisk-Users] Canreinvite cosa vedo a console -- Executing Dial(SIP/40001-3760, SIP/40002|30) in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760 -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message82.X2.XX3.X3 40002 146b518a4cd 00103/0 alaw No Tx: ACK 82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK2 active SIP channels Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta. ___--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
[Asterisk-Users] sip show inuse is useless!
Hi all, We have a SIP trunk with * and even when there are calls in progress sip show inuse always shows 0 calls in progress. I have outgoinglimit and incominglimit limit set and have also tried call-limit. sip show inuse works fine with SIP handsets though very frustrating. ___ --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] Echo Problem with T411P
Hi Steve, Thank you for your answers. First of all span 3 is not a satellite link and no echo occurs when I connect this line to another pbx with HW EC feature. I use TE411P with hardware EC and asterisk version 1.2.5. Do I have to do something to enable EC for this card ? Idris -Original Message- From: Steve Davies [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Echo Problem with T411P More than 128ms? 128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but definitely more than 16ms. No, 128ms = 1024 taps Like what sangoma offers. Ding, Ding, Ding, Ding! Okay, to be complete in my answers: No I do not get more than 128ms delay caused by European routing (I only threw that in as an example anyway), but asterisk's software cancellers only cancel 16ms, any more than that seems fairly buggy, and eats CPU. On the other hand, if that is a satellite link on span 3, you could easily get latency in excess of 1 second, which it should be the provider's responsibility to cancel, not the end user's IMHO. I also agree that the sangoma EC is excellent :) Do we know what E1/T1 hardware is in use here, and whether hardware EC is available? Steve ___ --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] Echo Problem with T411P
On 6/19/06, Idris AVCI [EMAIL PROTECTED] wrote: Hi Steve, Thank you for your answers. First of all span 3 is not a satellite link and no echo occurs when I connect this line to another pbx with HW EC feature. I use TE411P with hardware EC and asterisk version 1.2.5. Do I have to do something to enable EC for this card ? :) Now you've defeated me. I imagine that you need to do something to enable EC on that card, but it is not a card I know, so I'll leave it to someone who knows the card to offer any suggestions. Best of luck. Steve ___ --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 voicemail problem with isdn avm fritz!card
On Mon, 19 Jun 2006, Benjamin Sebbah wrote:
Hello everyone,
I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz,
256MB) with a TDM40b a TDM04b and an avm fritz!card pci.
I experience a problem with voicemail: my messages are good unless the
incoming call comes from isdn, which means via the avm fritz!card. In
this case (and in this case only) the message is disjointed and I can
hear at most 1 second out of a 1 minute message.
If the message comes from TDM400 then the message is perfect (even
though I still have a problem to detect the end of the call but that's
no big deal)
If the incoming call is answered (and not sent to voicemail because busy
or unavail) the sound is perfect.
I never heard of such a problem before. Can you please create a log of such
a call with
set verbose 9
capi debug
(might be big)
Armin
I hope you'll be able to help me.
Thanks
Benjamin SEBBAH
ADUNEO France
Here are my config files:
/etc/asterisk/capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=fr ;set default language
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
controller=1 ;capi controller number to use
group=9 ;dialout group
softdtmf=on ;enable/disable software dtmf detection, recommended
for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
echocancelold=yes;use facility selector 6 instead of correct 8
(necessary for older eicon drivers)
echotail=64 ;echo cancel tail setting
devices=2;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
and the interesting lines from /etc/asterisk/extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
PIERRE=Zap/1
MARC=SIP/marc
PATRICK=Zap/3
PROSPECT=Zap/2
OPENSPACE=Zap/4
FT_FREE=Zap/5
FT_ALICE=Zap/6
VOIP_FREE=Zap/7
VOIP_ALICE=Zap/8
NUMERIS=CAPI/ISDN1
[macro-repondeur]
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},15,rWw) ; Ring the interface,
15 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
;exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
;exten = s-BUSY,2,Goto(default,s,1) ; If they press #, return to
start
exten = _s-.,1,Goto(s-NOANSWER,1); Treat anything else
as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user
into VoicemailMain
[capi-in]
;standard: fait tout sonner
exten = 3090,1,Answer;
;exten = 3090,2,Macro(repondeur,8427,${OPENSPACE}${MARC}${PIERRE});
exten = 3090,2,Macro(repondeur,8427,${OPENSPACE}${PIERRE});
;Service technique
exten = 3091,1,Answer;
;exten = 3091,2,Macro(repondeur,3091,${OPENSPACE}${MARC});
exten = 3091,2,Macro(repondeur,3091,${OPENSPACE});
;Service commercial
exten = 3092,1,Answer;
exten = 3092,2,Macro(repondeur,3092,${PATRICK});
;Direction technique
exten = 3093,1,Answer;
;exten = 3093,2,Macro(repondeur,3093,${MARC});
exten = 3093,2,Macro(repondeur,3093,${OPENSPACE});
;non assigne pour le moment fait sonner uniquement le DECT
exten = 3094,1,Answer;
exten = 3094,2,Macro(repondeur,3094,${OPENSPACE});
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[Asterisk-Users] What type of T1 Cards to use for my Asterisk PBX
My Telco is bringing a T1 line to my company. It will be delivered via Copper. From my research on the net and in this group, I've found out that I have the following options: · OPTION 1 - Ensure that the My PBX Equipment (CPE) provides a T1 interface. · OPTION 2 - Convert the T1 into 24 standard analog lines. This would require channel bank. I use Asterisk, my existing system has 12 analog trunk lines. From the options mentioned above, 1) Are there any preferences with Option 1 vs Option 2? What do you recommend.? Thanks ---Dakota ___ --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] What type of T1 Cards to use for my Asterisk PBX
Hi Dakota, I think that you would have to opt to the first option, why T1 must be digital, where you he must have a TE110P inyour Asterisk. Of preference it opts to ISDN, therefore total it is supported in asterisk and much more easy of if programming. I wait to have helped GreetingsJosué 2006/6/19, [EMAIL PROTECTED] [EMAIL PROTECTED]: My Telco is bringing a T1 line to my company. It will be delivered viaCopper. From my research on the net and in this group, I've found out that I have the following options:·OPTION 1 - Ensure that the My PBX Equipment (CPE) provides a T1interface.·OPTION 2 - Convert the T1 into 24 standard analog lines. This wouldrequire channel bank. I use Asterisk, my existing system has 12 analog trunk lines. From theoptions mentioned above,1) Are there any preferences with Option 1 vs Option 2?What do you recommend.?Thanks---Dakota ___--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
[Asterisk-Users] Read command
Hi, Im using the Read command the read a DTMF tone. In this read command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede email: [EMAIL PROTECTED] internet: www.mobillion.nl ___ --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] Hitting * in a queue call hangs up?
Correct, And no, I am not passing H.This was identified as a bug in the chan_agents code. On 6/17/06, Wes Baehr [EMAIL PROTECTED] wrote: Create a context for your queue and put a '*' extension to redirect them back to the main menu (or wherever) Also, you're not passing option 'H' to Queue(), right? 'H' -- allow caller to hang up by hitting *. (I believe the actual hangup digit is defined by features.conf, but I could be wrong) Wes Baehr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Friday, June 16, 2006 4:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hitting * in a queue call hangs up? Kevin/et all, I thought this was mentioned in another thread, but I can't find it now.. Does 1.2.9.x fix this? If not.. what do I need to comment out to prevent * from hanging up on people when they come in a queue? On 6/12/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: - BJ Weschke [EMAIL PROTECTED] wrote: This was a hardcoded feature in Asterisk 1.2.X versions. It's now an optional feature in /trunk and will be going forward. And this is only true for queue members that are chan_agent agents. If you don't use chan_agent, you won't see this behavior either. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.0/368 - Release Date: 6/16/2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.0/368 - Release Date: 6/16/2006 ___ --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] multiple port
On 6/18/06, unplug [EMAIL PROTECTED] wrote: Hi, Does asterisk support mutl-port binding? Say beside setting the port 5060 in sip.conf, I want to use another port, say 6060. How can I set to use more than one port. Is it possible? unplug Not possible in Asterisk, but you should be able to do so with iptables by forwarding your secondary port to the primary port. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] free sun boxes
Greetings All, The Ultra 5 will take Solaris 10 no problem, however RAM will be an issue. Be sure that there is at least 128MB of RAM on these units or Solaris 10 will tend to chug. The SparcStation, from everything I've read, is not supported under Solaris 10. You can, however, get older versions of Solaris on it (7,8). Other than that the Ultra 5 was really the last desktop workstation that Sun produced that was truly workstation class. While they are returning to that with their new Opteron boxes, but the SunBlade series was a true embarrassment. In 5 years, I've lost a lot more Sunblade 100/150 boxes to lame stuff like motherboard failures than I've ever lost Ultra's. You should be working on your Ultra 5 for years to come. RandyW Angelito Manansala wrote: how much are you selling that stuff? On 6/18/06, Mike Fedyk [EMAIL PROTECTED] wrote: I'm in southern California, are you close or can you ship? Bob Knight wrote: I have 4 sparc based sun boxes I am about to pay money so I can get rid of them. They are running older versions of Solaris. You should be able to load Solaris 10 and play around with * on them. Time to clean the office: 3 Ultra 5 1 Sparcstation 5 I also have a box full of Sun keyboards and mice. Contact me offline if you want them. I've had many good years of development on them and it kills me to just toss them, but the office is just too damn cluttered. thanks, 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 ___ --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 voicemail problem with isdn avm fritz!card
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
Date: Monday, June 19, 2006 1:48 pm
Subject: Re: [Asterisk-Users] Asterisk voicemail problem with isdn avm
fritz!card
On Mon, 19 Jun 2006, Benjamin Sebbah wrote:
Hello everyone,
I have Asterisk SVN-trunk-r7498 installed on a server (celeron
2.4 Ghz,
256MB) with a TDM40b a TDM04b and an avm fritz!card pci.
I experience a problem with voicemail: my messages are good
unless the
incoming call comes from isdn, which means via the avm
fritz!card. In
this case (and in this case only) the message is disjointed and I
can hear at most 1 second out of a 1 minute message.
If the message comes from TDM400 then the message is perfect (even
though I still have a problem to detect the end of the call but
that's no big deal)
If the incoming call is answered (and not sent to voicemail
because busy
or unavail) the sound is perfect.
I never heard of such a problem before. Can you please create a log
of such
a call with
set verbose 9
capi debug
(might be big)
Armin
Actually I have just found a solution:
in capi.conf I've changed:
rxgain=0.8
txgain=0.8
echosquelch=1
echocancelold=yes
to
rxgain=1
txgain=0.8
echosquelch=2
echocancelold=no
and this works. Thanks for your help.
I hope you'll be able to help me.
Thanks
Benjamin SEBBAH
ADUNEO France
Here are my config files:
/etc/asterisk/capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=fr ;set default language
[ISDN1] ;this example interface gets name 'ISDN1' and
may be any
;name not starting with 'g' or 'contr'.
isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct
inward dial)
;when using NT-mode, 'DID' should be set in any
case incomingmsn=*;allow incoming calls to this list of
MSNs/DIDs, * = any
controller=1 ;capi controller number to use
group=9 ;dialout group
softdtmf=on ;enable/disable software dtmf detection,
recommended for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
echocancelold=yes;use facility selector 6 instead of correct 8
(necessary for older eicon drivers)
echotail=64 ;echo cancel tail setting
devices=2;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
and the interesting lines from /etc/asterisk/extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
PIERRE=Zap/1
MARC=SIP/marc
PATRICK=Zap/3
PROSPECT=Zap/2
OPENSPACE=Zap/4
FT_FREE=Zap/5
FT_ALICE=Zap/6
VOIP_FREE=Zap/7
VOIP_ALICE=Zap/8
NUMERIS=CAPI/ISDN1
[macro-repondeur]
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here
as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},15,rWw) ; Ring the
interface, 15 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
;exten = s-NOANSWER,2,Goto(default,s,1); If they press
#, return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
voicemail w/
busy announce
;exten = s-BUSY,2,Goto(default,s,1); If they press #,
return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *,
send the user
into VoicemailMain
[capi-in]
;standard: fait tout sonner
exten = 3090,1,Answer;
;exten =
3090,2,Macro(repondeur,8427,${OPENSPACE}${MARC}${PIERRE}); exten
= 3090,2,Macro(repondeur,8427,${OPENSPACE}${PIERRE});
;Service technique
exten = 3091,1,Answer;
;exten = 3091,2,Macro(repondeur,3091,${OPENSPACE}${MARC});
exten = 3091,2,Macro(repondeur,3091,${OPENSPACE});
;Service commercial
exten = 3092,1,Answer;
exten = 3092,2,Macro(repondeur,3092,${PATRICK});
;Direction technique
exten = 3093,1,Answer;
;exten = 3093,2,Macro(repondeur,3093,${MARC});
exten = 3093,2,Macro(repondeur,3093,${OPENSPACE});
;non assigne pour le moment fait sonner uniquement le DECT
exten = 3094,1,Answer;
exten = 3094,2,Macro(repondeur,3094,${OPENSPACE});
___
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Re: [Asterisk-Users] sip show inuse is useless!
What version of * are you using? I am running 1.2.7.1 with call-limit= and it works fine. bp On 6/19/06, Eric Bishop [EMAIL PROTECTED] wrote: Hi all,We have a SIP trunk with * and even when there are calls in progress sip show inuse always shows 0 calls in progress. I have outgoinglimit and incominglimit limit set and have also tried call-limit. sip show inuse works fine with SIP handsets though very frustrating. ___--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] sip show inuse is useless!
I have tried it with 1.2.7.1 and 1.2.9.1. Same issue with both and only on the SIP trunk, not on endpoints.On 6/19/06, William Piper [EMAIL PROTECTED] wrote: What version of * are you using? I am running 1.2.7.1 with call-limit= and it works fine. bp On 6/19/06, Eric Bishop [EMAIL PROTECTED] wrote: Hi all,We have a SIP trunk with * and even when there are calls in progress sip show inuse always shows 0 calls in progress. I have outgoinglimit and incominglimit limit set and have also tried call-limit. sip show inuse works fine with SIP handsets though very frustrating. ___--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 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] Which phones are good, or at least acceptable, for home and office
I liked the ringer that read the phone number too, but a couple months ago, I did a firmware upgrade, and that ringer option went away Do you have the latest firmware?? I upgraded because of a problem with my phone losing registration, which is now fixed, but I lost that really cool feature Any idea how to get that back? -Steve From: Lacy Moore - Aspendora [mailto:[EMAIL PROTECTED] Sent: Saturday, June 17, 2006 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Which phones are good, or at least acceptable,for home and office The Grandstream seem to be a crap shoot. Some people have real good luck, others don't. So far, I've got four of them in use and the users seem to be happy. The only drawback that I have is that there is no way I can even attempt to try to explain the complex method that you have to use to PARK a call. Their attended transfers are weird. I really like the ringer that calls out the caller ID. It's because of that, that I might put them in my house. However, I still have a CIDCO device that reads out the caller ID. My house is small enough that I can hear it all over the house. I would also like to try out the Aastra 9133. It's a little more than the GXP2000. And, I have noticed the handset gets warm on the GXP. Others have mentioned this. For more information, including things already discussed about the Grandstreams, you can try: http://www.google.com/search?hl=enlr=q=site%3Ahttp%3A%2F%2Flists.digium.com%2Fpipermail%2Fasterisk-users%2F+grandstream or This site http://www.asteriskguru.com/archives/asterisk-users-vf2.html?sid=d6b13ed5fdbe515037bc9738c24f contains a complete archive of this list in forum format. ___ --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 voicemail problem with isdn avm fritz!card
On Mon, 19 Jun 2006, Benjamin Sebbah wrote:
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
Date: Monday, June 19, 2006 1:48 pm
Subject: Re: [Asterisk-Users] Asterisk voicemail problem with isdn avm
fritz!card
On Mon, 19 Jun 2006, Benjamin Sebbah wrote:
Hello everyone,
I have Asterisk SVN-trunk-r7498 installed on a server (celeron
2.4 Ghz,
256MB) with a TDM40b a TDM04b and an avm fritz!card pci.
I experience a problem with voicemail: my messages are good
unless the
incoming call comes from isdn, which means via the avm
fritz!card. In
this case (and in this case only) the message is disjointed and I
can hear at most 1 second out of a 1 minute message.
If the message comes from TDM400 then the message is perfect (even
though I still have a problem to detect the end of the call but
that's no big deal)
If the incoming call is answered (and not sent to voicemail
because busy
or unavail) the sound is perfect.
I never heard of such a problem before. Can you please create a log
of such
a call with
set verbose 9
capi debug
(might be big)
Armin
Actually I have just found a solution:
in capi.conf I've changed:
rxgain=0.8
txgain=0.8
echosquelch=1
echocancelold=yes
to
rxgain=1
txgain=0.8
echosquelch=2
echocancelold=no
and this works. Thanks for your help.
Ah, sure. I think it's just the echosquelch setting. echocancelold applies
for Eicon cards only and echosquelch causes frame-length changes.
Armin
I hope you'll be able to help me.
Thanks
Benjamin SEBBAH
ADUNEO France
Here are my config files:
/etc/asterisk/capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=fr ;set default language
[ISDN1] ;this example interface gets name 'ISDN1' and
may be any
;name not starting with 'g' or 'contr'.
isdnmode=DID ;'MSN' (point-to-multipoint) or 'DID' (direct
inward dial)
;when using NT-mode, 'DID' should be set in any
case incomingmsn=*;allow incoming calls to this list of
MSNs/DIDs, * = any
controller=1 ;capi controller number to use
group=9 ;dialout group
softdtmf=on ;enable/disable software dtmf detection,
recommended for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
echocancelold=yes;use facility selector 6 instead of correct 8
(necessary for older eicon drivers)
echotail=64 ;echo cancel tail setting
devices=2;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
and the interesting lines from /etc/asterisk/extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
PIERRE=Zap/1
MARC=SIP/marc
PATRICK=Zap/3
PROSPECT=Zap/2
OPENSPACE=Zap/4
FT_FREE=Zap/5
FT_ALICE=Zap/6
VOIP_FREE=Zap/7
VOIP_ALICE=Zap/8
NUMERIS=CAPI/ISDN1
[macro-repondeur]
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here
as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},15,rWw) ; Ring the
interface, 15 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
;exten = s-NOANSWER,2,Goto(default,s,1) ; If they press
#, return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
voicemail w/
busy announce
;exten = s-BUSY,2,Goto(default,s,1) ; If they press #,
return to start
exten = _s-.,1,Goto(s-NOANSWER,1); Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *,
send the user
into VoicemailMain
[capi-in]
;standard: fait tout sonner
exten = 3090,1,Answer;
;exten =
3090,2,Macro(repondeur,8427,${OPENSPACE}${MARC}${PIERRE}); exten
= 3090,2,Macro(repondeur,8427,${OPENSPACE}${PIERRE});
;Service technique
exten = 3091,1,Answer;
;exten = 3091,2,Macro(repondeur,3091,${OPENSPACE}${MARC});
exten = 3091,2,Macro(repondeur,3091,${OPENSPACE});
;Service commercial
exten = 3092,1,Answer;
exten = 3092,2,Macro(repondeur,3092,${PATRICK});
;Direction technique
exten = 3093,1,Answer;
;exten = 3093,2,Macro(repondeur,3093,${MARC});
exten = 3093,2,Macro(repondeur,3093,${OPENSPACE});
;non assigne pour le moment fait sonner uniquement le DECT
exten =
Re: [Asterisk-Users] Transfer call via AMI or dialplan
Piece of cake Julian: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect Regards On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: At the moment when one of our users wants to transfer a call, they press the transfer button on the phone, enter the extension and away they go. Is there any way to do this via the AMI or dialplan ? I want them to push a button on the screen rather than using the phone itself. Julian ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Transfer call via AMI or dialplan
If you know which channel you want to transfer, then one way is to use the Redirect AMI action (http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Actio n+Redirect). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Monday, June 19, 2006 9:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Transfer call via AMI or dialplan At the moment when one of our users wants to transfer a call, they press the transfer button on the phone, enter the extension and away they go. Is there any way to do this via the AMI or dialplan ? I want them to push a button on the screen rather than using the phone itself. Julian ___ --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] What ever happened to the LTAPI, the Linux Telephony API?
On Sunday 18 June 2006 13:38, Brian Capouch wrote: The hardware is pretty crappy. I never had any real trouble with the QuickNet PhoneJack PCI cards (I have three, one I blew out the SLIC because I hooked it up to POTS and someone rang me), but then again I haven't touched them in probably two years now. :-) -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
[Asterisk-Users] Bristuff-0.3.0-PRE-1q and florz patch compile trouble
Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a x86_64 box (I guess nobody is using x86_64 platform or is able to fix this themselves?) First of all when bristuff is downloaded and compile is started it appears that the bristuff Makefiles are badly broken. The asterisk Makefiles all do see to find the kernel sources on a RHEL4 box in the proper directory, the pure bristuff things break because they expect the kernel sources *only* in /usr/src/kernel-2.6 OK, so I created a symlink to make bristuff happy Zaptel does compile, so does libpri but not Junghanns GSM stuff and not quadbri and also not cwain (which I don't really care for). However zaphfc doesn't compile either, I get the old error again: rm -f zaphfc.o *.ko *.mod.c *.mod.o .*o.cmd *~ rm -rf .tmp_versions Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 install -D -m 644 zaphfc.ko /lib/modules/`uname -r`/misc/zaphfc.ko install: cannot stat `zaphfc.ko': No such file or directory make: *** [installlinux26] Error 1 hfc-pci driver installed. Press Enter to continue, or CTRL + C to abort. In the past I used to 'fix' this by modifying the Make file. After applying florz patch previously I had to modify KSRC=/usr/src/linux to /usr/src/linux-2.6 but with the Makefile of zaptel 1.2.5 this line is gone??? (I used Zaptel 1.1x previously) (And yes in spite of the above error /usr/src/linux-2.6 *is* linked properly to my kernel source) Help Thanks for any hints / tips ___ --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] suggestions for Wireless phone that receives text messages
I am looking for wireless SIP phones that will also receive a text message. Has anyone phone such a phone? Thanks, jerry ___ --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] Setting caller-id when parking call
I have an issue where someone will park a call, and then it will ring back to them, but because the caller-id looks like a regular inbound call, they don't know how to answer the call (these are the receptionists). I've tried to make an extention that I can transfer to that will set the caller-id, and that works, but I'm having issues. For instance if I do a blind transfer to an extention the CALLER hears '71' instead of the receptionist.. DOH! That didn't work. If I do an attenteded transfer, then the caller-id gets set, but when the person hangs up, it gets unset. DOH! That didn't work either now what? :) ___ --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] WRTG54GS Capacity
Given that the NSLU2 can't do trunking, do you think that a PIII 733Mhz, 128MB RAM will do? Thanks, Daniel On Jun 15, 2006, at 4:15 AM, Tim Panton wrote: On 15 Jun 2006, at 02:59, Daniel Salama wrote: Does anyone know how many simultaneous calls can a WRTG54GS handle? Assuming SIP phones are connected locally using G711.u codec and the WRTG54GS connects to a remote Asterisk server using IAX2 trunking using GSM codec. Very few (2 perhaps) - You will be transcoding on the WRTG54. On that sort of box you need to stick to a single codec. In your case I guess GSM. If you want to transcode, you will need a bigger cpu. If your phones support it, I'd use GSM everywhere, since your original problem was bandwidth. Do take a look at the OpenSlug on the nslu2 - The nice thing about the 'Slug' is that you can add a USB harddrive for swap and voicemail, so it is more 'expandable' than the WRTG54 I should warn you I have never tried trunking IAX on my slug, I will do at some point Tim. Tim Panton [EMAIL PROTECTED] ___ --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] DTMF Talk off
Check
http://lists.digium.com/pipermail/asterisk-users/2005-January/078141.html
On Sun, 18 Jun 2006, John Millican wrote:
Hello all,
I have seen some chatter again about DTMF. I see most of the talk about DTMF
around not being able to get an external IVR to recognize digits, not a big
issue for me at this time but sill interesting. My issue though, is with
talk off on a zap channel. It seems to be getting worse or maybe my patience
is thinning. All my calls go out and come in pstn through an FXO as I do not
have high speed available here at home. My Current setup is:
Phone--PAP2-- * ---PSTN---Voip number to * at another location(that has
high speed)---send to VoIP provider
I read a post about talked about the length of the DTMFish sound. I also
remeber seing something about relaxdtmf being set to something other than yes
or no, so I looked in chan_zap.c and found relaxdtmf in many places but it
looked to my inexperienced eye that it could only be set to 'yes' or 'no',
but i did find some mention of tonlength (at line 10858)
followed that to zaptel.c (line 3357) where it said :
if ((tdp.dtmf_tonelen 4000 ) || (tdp.dtmf_tonelen 10 ))
return -EINVAL
Which I am guessing means unless the dtmf is between these 2 values ignore it.
Any ideas what might happen if i increased the minimum time value? if this is
indeed what this is referring to?
Zapata.conf:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
busydetect=yes
busycount=6
echocancel=128
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=0
immediate=no
context=default
signalling=fxs_ks
channel = 1
same for channel 2
zaptel.conf:
loadzone = us
fxsks=1
fxsks=2
extensions.conf:
exten = s,1, NoOp(${CALLERID} time ${DATETIME});
exten = s,2, Dial(sip/677sip/666,30,tT);
exten = a bunch of stuff to do with agi look ups and voicemail
leave/retrieve
All very basic and works like a charm except for the talk off.
Thanks in advance to all,
John M
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deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
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[Asterisk-Users] How to use a data T-1?
I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren ___ --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] How to use a data T-1?
Warren, My suggestion for testing would be just use ethernet hand off to the asterisk from the Cisco. You could bypass the Cisco but then you would need a T-1 card for the asterisk box and they are not cheap. I believe there are valid arguments for both choices though and ultimately should be decided by what you are planning as a final solution. John M On Monday June 19 2006 10:15 am, Warren wrote: I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren ___ --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] How to use a data T-1?
You don't need a T1 card for a data T1. Just run it through your Cisco box send it over to your NIC on the asterisk box. bp On 6/19/06, Warren [EMAIL PROTECTED] wrote: I have a data T-1 available to me to do some testing of a new asterisksystemthat I am putting together.Do I just leave this T routed through my cisco router and plug in the asterisk system through a network cardor do I need to get a T-1 card and use that?I looked on the voip-infowiki and it did not seem to answer this for me.TIA,Warren ___--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] How to use a data T-1?
I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren If this data T-1 just goes to the Internet then you would use it just like any other network connection at your cisco router. If this data T-1 goes between two sites of yours then you could use it either as a dedicated route between network cards on each end (that connect to cisco or other brand routers) or a voice route between two Asterisk servers with voice T-1 cards. The choice would be between capacity for say G729 trunks over a data link or latency as voice T-1s. -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --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] How to use a data T-1?
John, Thanks for the quick reply. I do intend to get a T-1 card anyway. Would it be the same card for a data T-1 as for a voice T-1 just with different setup? W John Millican wrote: Warren, My suggestion for testing would be just use ethernet hand off to the asterisk from the Cisco. You could bypass the Cisco but then you would need a T-1 card for the asterisk box and they are not cheap. I believe there are valid arguments for both choices though and ultimately should be decided by what you are planning as a final solution. John M On Monday June 19 2006 10:15 am, Warren wrote: I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren ___ --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] How to use a data T-1?
Depends what you want to do! Do you want to do VoIP over that T1 to a provider or IP telephones? Do you want to hook up to the PSTN through that T1 as 24 voice channels, through a T1 card on your asterisk? If you want to use the T1 as 24 voice channels, the Telco is going to have to re-provision the T1 as a voice T1, because currently, presumably it is one big channel of data. You could have the telco do any combination of 24 channels, some voice and some data, if your DSU or router allows drop and insert of channels. It would then split the T1 into a voice side and a data side, each with part of the channels available. Once you have a channelized voice T1, it can plug into a voice T1 card in your Asterisk, but typically can't do data anymore, so if that's not what you intend, then please explain further.. -Original Message- From: Warren [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How to use a data T-1? I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren ___ --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] Bristuff-0.3.0-PRE-1q and florz patch compile trouble
On Mon, Jun 19, 2006 at 03:41:30PM +0200, Remco Barendse wrote: Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a x86_64 box (I guess nobody is using x86_64 platform or is able to fix this themselves?) First of all when bristuff is downloaded and compile is started it appears that the bristuff Makefiles are badly broken. The asterisk Makefiles all do see to find the kernel sources on a RHEL4 box in the proper directory, the pure bristuff things break because they expect the kernel sources *only* in /usr/src/kernel-2.6 OK, so I created a symlink to make bristuff happy Zaptel does compile, so does libpri but not Junghanns GSM stuff and not quadbri and also not cwain (which I don't really care for). However zaphfc doesn't compile either, I get the old error again: rm -f zaphfc.o *.ko *.mod.c *.mod.o .*o.cmd *~ rm -rf .tmp_versions Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 install -D -m 644 zaphfc.ko /lib/modules/`uname -r`/misc/zaphfc.ko install: cannot stat `zaphfc.ko': No such file or directory make: *** [installlinux26] Error 1 hfc-pci driver installed. Press Enter to continue, or CTRL + C to abort. In the past I used to 'fix' this by modifying the Make file. After applying florz patch previously I had to modify KSRC=/usr/src/linux to /usr/src/linux-2.6 but with the Makefile of zaptel 1.2.5 this line is gone??? (I used Zaptel 1.1x previously) (And yes in spite of the above error /usr/src/linux-2.6 *is* linked properly to my kernel source) The bristuff modules makefile replicates some functionality unnecessarily. Add the following to zaptel's Makefile: MODULES+=zaphfc (after the line 'MODULES+=ztdummy') Copy zaphfc.c and zaphfc.h to the zaptel directory, and install zaptel according to the standard instructions for building zaptel. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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] DTMF Talk off
John,
Well I am certainly not an expert on this. I am using an SPA-3000 and I
have not experienced this. I did have to go to inband on the fxo channel
as rfc8322 did not work for ivr's when using Asterisk. I think you said
you were using a linksys or sipura product for you fxo?? If that is the
case using inband and the ulaw/alaw encoder for the fxo channel might
help. Worth a try I guess. There are some rfc8322 issues that apparently
will be addressed with a rewrite in the next makor version release.
Doug
On Mon, 19 Jun 2006, John Millican wrote:
Doug, I read that post and unfortunately it was not a solution. I do not
believe it has to do with interstate as it happens intra state also. Is
there any way to make DTMF detection stricter, ie require a longer minimum
tone length. Assuming ( yes a dangerous practice) that the human voice will
not hold a DTMF sequence stable for very long, if I lengthen the minimum
required length I may be able to minimize the talk off. What do you think?
Any suggestions?
John M
Doug Crompton wrote:
Check
http://lists.digium.com/pipermail/asterisk-users/2005-January/078141.html
On Sun, 18 Jun 2006, John Millican wrote:
Hello all,
I have seen some chatter again about DTMF. I see most of the talk about
DTMF around not being able to get an external IVR to recognize digits,
not a big issue for me at this time but sill interesting. My issue
though, is with talk off on a zap channel. It seems to be getting worse
or maybe my patience is thinning. All my calls go out and come in pstn
through an FXO as I do not have high speed available here at home. My
Current setup is:
Phone--PAP2-- * ---PSTN---Voip number to * at another location(that
has high speed)---send to VoIP provider
I read a post about talked about the length of the DTMFish sound. I also
remeber seing something about relaxdtmf being set to something other than
yes or no, so I looked in chan_zap.c and found relaxdtmf in many places
but it looked to my inexperienced eye that it could only be set to 'yes'
or 'no', but i did find some mention of tonlength (at line 10858)
followed that to zaptel.c (line 3357) where it said :
if ((tdp.dtmf_tonelen 4000 ) || (tdp.dtmf_tonelen 10 ))
return -EINVAL
Which I am guessing means unless the dtmf is between these 2 values
ignore it. Any ideas what might happen if i increased the minimum time
value? if this is indeed what this is referring to?
Zapata.conf:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
busydetect=yes
busycount=6
echocancel=128
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=0
immediate=no
context=default
signalling=fxs_ks
channel = 1
same for channel 2
zaptel.conf:
loadzone = us
fxsks=1
fxsks=2
extensions.conf:
exten = s,1, NoOp(${CALLERID} time ${DATETIME});
exten = s,2, Dial(sip/677sip/666,30,tT);
exten = a bunch of stuff to do with agi look ups and voicemail
leave/retrieve
All very basic and works like a charm except for the talk off.
Thanks in advance to all,
John M
Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954 *
* 215-431-6307*
* *
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
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Re: [Asterisk-Users] How to use a data T-1?
Warren, Yes. The setup is based on what type of signaling the telco is giving you. John On Monday June 19 2006 10:32 am, Warren wrote: John, Thanks for the quick reply. I do intend to get a T-1 card anyway. Would it be the same card for a data T-1 as for a voice T-1 just with different setup? W John Millican wrote: Warren, My suggestion for testing would be just use ethernet hand off to the asterisk from the Cisco. You could bypass the Cisco but then you would need a T-1 card for the asterisk box and they are not cheap. I believe there are valid arguments for both choices though and ultimately should be decided by what you are planning as a final solution. John M On Monday June 19 2006 10:15 am, Warren wrote: I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren ___ --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] DTMF Talk off
Doug,
thanks for the help. I am using uLAW and inband every where. I have tried
using 2833 and it did not appear to make any difference, better or worse.
this is why I was thinking that if I could increase the minimum required time
for a tone that it night help, I am just not sure where the best place top do
this is. i thought I had seen a post about setting relaxdtmf to a value to
actually make dtmf detection stricter but i can not seam to find anything
other than 'yes' or 'no'.
John
Doug Crompton wrote:
John,
Well I am certainly not an expert on this. I am using an SPA-3000 and I
have not experienced this. I did have to go to inband on the fxo channel
as rfc8322 did not work for ivr's when using Asterisk. I think you said
you were using a linksys or sipura product for you fxo?? If that is the
case using inband and the ulaw/alaw encoder for the fxo channel might
help. Worth a try I guess. There are some rfc8322 issues that apparently
will be addressed with a rewrite in the next makor version release.
Doug
On Mon, 19 Jun 2006, John Millican wrote:
Doug, I read that post and unfortunately it was not a solution. I do not
believe it has to do with interstate as it happens intra state also. Is
there any way to make DTMF detection stricter, ie require a longer
minimum tone length. Assuming ( yes a dangerous practice) that the human
voice will not hold a DTMF sequence stable for very long, if I lengthen
the minimum required length I may be able to minimize the talk off. What
do you think? Any suggestions?
John M
Doug Crompton wrote:
Check
http://lists.digium.com/pipermail/asterisk-users/2005-January/078141.ht
ml
On Sun, 18 Jun 2006, John Millican wrote:
Hello all,
I have seen some chatter again about DTMF. I see most of the talk
about DTMF around not being able to get an external IVR to recognize
digits, not a big issue for me at this time but sill interesting. My
issue though, is with talk off on a zap channel. It seems to be
getting worse or maybe my patience is thinning. All my calls go out
and come in pstn through an FXO as I do not have high speed available
here at home. My Current setup is:
Phone--PAP2-- * ---PSTN---Voip number to * at another
location(that has high speed)---send to VoIP provider
I read a post about talked about the length of the DTMFish sound. I
also remeber seing something about relaxdtmf being set to something
other than yes or no, so I looked in chan_zap.c and found relaxdtmf
in many places but it looked to my inexperienced eye that it could
only be set to 'yes' or 'no', but i did find some mention of
tonlength (at line 10858) followed that to zaptel.c (line 3357) where
it said :
if ((tdp.dtmf_tonelen 4000 ) || (tdp.dtmf_tonelen 10 ))
return -EINVAL
Which I am guessing means unless the dtmf is between these 2 values
ignore it. Any ideas what might happen if i increased the minimum
time value? if this is indeed what this is referring to?
Zapata.conf:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
busydetect=yes
busycount=6
echocancel=128
echocancelwhenbridged=yes
echotraining=yes
rxgain=0
txgain=0
immediate=no
context=default
signalling=fxs_ks
channel = 1
same for channel 2
zaptel.conf:
loadzone = us
fxsks=1
fxsks=2
extensions.conf:
exten = s,1, NoOp(${CALLERID} time ${DATETIME});
exten = s,2, Dial(sip/677sip/666,30,tT);
exten = a bunch of stuff to do with agi look ups and voicemail
leave/retrieve
All very basic and works like a charm except for the talk off.
Thanks in advance to all,
John M
Those that sacrifice essential liberty to obtain a little temporary safety
deserve neither liberty nor safety. -- Ben Franklin (1759)
* Doug Crompton *
* Richboro, PA 18954*
* 215-431-6307 *
**
* [EMAIL PROTECTED]*
* http://www.crompton.com *
___
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[Asterisk-Users] Meetme Dumping Call's
I have looked thru the users lists, google and voip-info and did not find an answer. I am using asterisk (latest SVN as of three weeks ago). I am using real time. I have a problem that when a user enters an invalid meetme extension the meetme says invalid room and dumps the call. Here is what I have in mysql:+-+--+---+--+--++| id | context | exten | priority | app | appdata |+-+--+---+--+--++| 155 | citicomco-internal | _5XXX | 1 | MeetMe | ${EXTEN}|cMrpsq || 285 |
citicomco-ivr-extens | _5XXX | 3 | Goto | citicomco-incoming|s|1 || 283 | citicomco-ivr-extens | _5XXX | 1 | MeetMe | ${EXTEN}|cMrpsq || 284 | citicomco-ivr-extens | _5XXX | 2 | Playback | goodbye || 481 | citicomco-internal | _5XXX | 2 | Goto | citicomco-incoming|s|1 |+-+--+---+--+--++
Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___
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Re: [Asterisk-Users] How to use a data T-1?
Steve, I want to end up with a system that will let me send and receive voice calls. I guess what I want to do depends on the best way to do that. Can I do more than 23 (decent sounding) voice calls on a data T-1 with someone else handling the final part of the call to the copper for me? If so than that is my likely final destination. I have a channelized voice T-1 currently plugged into my meridian system, but I would like (if realistically possible) to do as much of this over IP as possible for maximum flexibility. Is that a pipe dream or just silly given the current state of technology? I am lucky enough to work for a company that is letting me take my time with this, test the various options and come up with the proper solution. I am assuming (I know: dumb to assume) at this point that VoIP over a T-1 to a provider that can then route it to hard phones for me would be the way to go. Similarly, I would point my 800 number to a DiD hosted by a VoIP provider that would then route the call back to me. If that is an incorrect assumption, please let me know. Regards, Warren Steve Jones wrote: Depends what you want to do! Do you want to do VoIP over that T1 to a provider or IP telephones? Do you want to hook up to the PSTN through that T1 as 24 voice channels, through a T1 card on your asterisk? If you want to use the T1 as 24 voice channels, the Telco is going to have to re-provision the T1 as a voice T1, because currently, presumably it is one big channel of data. You could have the telco do any combination of 24 channels, some voice and some data, if your DSU or router allows drop and insert of channels. It would then split the T1 into a voice side and a data side, each with part of the channels available. Once you have a channelized voice T1, it can plug into a voice T1 card in your Asterisk, but typically can't do data anymore, so if that's not what you intend, then please explain further.. -Original Message- From: Warren [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How to use a data T-1? I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren ___ --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] How to use a data T-1?
Is anyone using the HDLC facility in Zaptel to bring a data T1 into an Asterisk system? I know this was available in kernel 2.4.19--is anyone using it in kernel 2.6.x? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --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 1.07 crash under Debian Sarge
I have just finished implementing an Asterisk system for my place of
business (first one), and after three days of flawless usage, Asterisk
seems to have crashed. I wasn't running with '-g', so I don't have a
core dump. Here's the sequence of events leading up to the crash:
1. call comes in on our TDM2400P
2. all of our phones (about 26 Polycoms) ring. (it's after biz.
hours, so all phones ring)
3. an employee answers the call.
4. the employee attempts a page (autoanswer + meetme AGI script with
Polycoms)
5. about half the phones make it to the meeting, then the system
crashes.
6. an executive calls my manager, who's on vacation, my manager calls
me, autopsy begins.
here's a few important snippets:
===extensions.conf=
[system-page]
exten = 999,1,Macro(system-page,${CALLERIDNUM})
; The first variable is the originating caller, the others are phones I
; wish to exclude from the system-wide paging.
[macro-system-page]
exten = s,1,AGI(allpage.agi|SIP/${CALLERIDNUM});@TODO make more
robust, not only SIP
exten = s,2,MeetMe(999,Adqt)
;exten = s,2,Hangup
[add-to-page]
exten = listener,1,MeetMe(999,dmqx)
===
==/var/log/asterisk/debug==
Jun 12 17:44:12 DEBUG[17975]: Building dynamic conference '999'
Jun 12 17:44:12 DEBUG[17975]: Placed channel SIP/302-6188 in ZAP conf
1023
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
...
Jun 12 17:44:18 DEBUG[17975]: Hangup: channel: -2 index = 0, normal =
51, callwait = -1, thirdcall = -1
Jun 12 17:44:18 DEBUG[17975]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-1321090091
Jun 12 17:44:18 DEBUG[17975]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:19 DEBUG[17975]: update_user_counter(302) - decrement inUse
counter
Jun 12 17:44:19 DEBUG[18016]: Building dynamic conference '999'
Jun 12 17:44:20 DEBUG[18016]: Placed channel SIP/508-af01 in ZAP conf
1023
Jun 12 17:44:20 DEBUG[18016]: Hangup: channel: -2 index = 0, normal =
41, callwait = -1, thirdcall = -1
Jun 12 17:44:20 DEBUG[18016]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-1583015986
Jun 12 17:44:20 DEBUG[18016]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:21 DEBUG[18016]: update_user_counter(508) - decrement
outUse counter
Jun 12 17:44:21 DEBUG[23992]: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 103: Found
Jun 12 17:44:21 DEBUG[18017]: Building dynamic conference '999'
Jun 12 17:44:22 DEBUG[18017]: Placed channel SIP/804-677b in ZAP conf
1023
Jun 12 17:44:22 DEBUG[18017]: Hangup: channel: -2 index = 0, normal =
41, callwait = -1, thirdcall = -1
Jun 12 17:44:22 DEBUG[18017]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-1132503448
Jun 12 17:44:22 DEBUG[18017]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:23 DEBUG[18017]: update_user_counter(804) - decrement
outUse counter
...
Jun 12 17:44:32 DEBUG[18041]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18019]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18021]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18028]: update_user_counter(404) - decrement
outUse counter
Jun 12 17:44:32 DEBUG[18042]: Placed channel SIP/401-1bec in ZAP conf
1023
Jun 12 17:44:32 DEBUG[18043]: Placed channel SIP/601-d011 in ZAP conf
1023
Jun 12 17:44:32 DEBUG[18043]: Hangup: channel: -2 index = 0, normal =
41, callwait = -1, thirdcall = -1
Jun 12 17:44:32 DEBUG[18043]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-726361999
Jun 12 17:44:32 DEBUG[18043]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:32 DEBUG[18041]: Placed channel SIP/203-6116 in ZAP conf
1023
CRASH
==
==/var/log/asterisk/messages==
Jun 12 17:40:49 WARNING[17955]: No such host: 806
Jun 12 17:40:49 NOTICE[17955]: Unable to create channel of type 'SIP'
Jun 12 17:40:53 WARNING[17955]: Unable to request echo training on
channel 1
Jun 12 17:43:42 WARNING[17958]: No such host: 806
Jun 12 17:43:42 NOTICE[17958]: Unable to create channel of type 'SIP'
Jun 12 17:43:44 WARNING[17958]: Unable to request echo training on
channel 1
Jun 12 17:44:12 NOTICE[18001]: Unable to request channel SIP/595
Jun 12 17:44:12 NOTICE[18004]: Unable to request channel SIP/808
Jun 12 17:44:12 NOTICE[18008]: Unable to request channel SIP/201
Jun 12 17:44:12 NOTICE[18011]: Unable to request channel SIP/212
Jun 12 17:44:12 NOTICE[17980]: Unable to request channel SIP/704
Jun 12 17:44:12 NOTICE[17984]: Unable to request channel SIP/802
Jun 12 17:44:12 NOTICE[17982]: Unable to request channel SIP/803
Jun 12 17:44:12 NOTICE[17985]: Unable to request channel SIP/801
Jun 12 17:44:32 WARNING[18041]: Conference not found
CRASH
RE: [Asterisk-Users] How to use a data T-1?
If you get it figured out, please post details on the wiki. I tried about a year ago. I think I was close but I didn't have enough time to pursue it. Looks to be trivial with Sangoma though I haven't tried that either. Thanks, Steve Totaro -Original Message- From: Michael Welter [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 11:15 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? Is anyone using the HDLC facility in Zaptel to bring a data T1 into an Asterisk system? I know this was available in kernel 2.4.19--is anyone using it in kernel 2.6.x? -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --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 1.07 crash under Debian Sarge
I suspect that the majority of the advice that you are going to get
would be to upgrade to the latest version of asterisk, as so many
changes and bug fixes have been made since the 1.07 release.
Julian.
Mark W. Stoddard wrote:
I have just finished implementing an Asterisk system for my place of
business (first one), and after three days of flawless usage, Asterisk
seems to have crashed. I wasn't running with '-g', so I don't have a
core dump. Here's the sequence of events leading up to the crash:
1. call comes in on our TDM2400P
2. all of our phones (about 26 Polycoms) ring. (it's after biz.
hours, so all phones ring)
3. an employee answers the call.
4. the employee attempts a page (autoanswer + meetme AGI script with
Polycoms)
5. about half the phones make it to the meeting, then the system
crashes.
6. an executive calls my manager, who's on vacation, my manager calls
me, autopsy begins.
here's a few important snippets:
===extensions.conf=
[system-page]
exten = 999,1,Macro(system-page,${CALLERIDNUM})
; The first variable is the originating caller, the others are phones I
; wish to exclude from the system-wide paging.
[macro-system-page]
exten = s,1,AGI(allpage.agi|SIP/${CALLERIDNUM});@TODO make more
robust, not only SIP
exten = s,2,MeetMe(999,Adqt)
;exten = s,2,Hangup
[add-to-page]
exten = listener,1,MeetMe(999,dmqx)
===
==/var/log/asterisk/debug==
Jun 12 17:44:12 DEBUG[17975]: Building dynamic conference '999'
Jun 12 17:44:12 DEBUG[17975]: Placed channel SIP/302-6188 in ZAP conf
1023
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
...
Jun 12 17:44:18 DEBUG[17975]: Hangup: channel: -2 index = 0, normal =
51, callwait = -1, thirdcall = -1
Jun 12 17:44:18 DEBUG[17975]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-1321090091
Jun 12 17:44:18 DEBUG[17975]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:19 DEBUG[17975]: update_user_counter(302) - decrement inUse
counter
Jun 12 17:44:19 DEBUG[18016]: Building dynamic conference '999'
Jun 12 17:44:20 DEBUG[18016]: Placed channel SIP/508-af01 in ZAP conf
1023
Jun 12 17:44:20 DEBUG[18016]: Hangup: channel: -2 index = 0, normal =
41, callwait = -1, thirdcall = -1
Jun 12 17:44:20 DEBUG[18016]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-1583015986
Jun 12 17:44:20 DEBUG[18016]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:21 DEBUG[18016]: update_user_counter(508) - decrement
outUse counter
Jun 12 17:44:21 DEBUG[23992]: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 103: Found
Jun 12 17:44:21 DEBUG[18017]: Building dynamic conference '999'
Jun 12 17:44:22 DEBUG[18017]: Placed channel SIP/804-677b in ZAP conf
1023
Jun 12 17:44:22 DEBUG[18017]: Hangup: channel: -2 index = 0, normal =
41, callwait = -1, thirdcall = -1
Jun 12 17:44:22 DEBUG[18017]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-1132503448
Jun 12 17:44:22 DEBUG[18017]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:23 DEBUG[18017]: update_user_counter(804) - decrement
outUse counter
...
Jun 12 17:44:32 DEBUG[18041]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18019]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18021]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18028]: update_user_counter(404) - decrement
outUse counter
Jun 12 17:44:32 DEBUG[18042]: Placed channel SIP/401-1bec in ZAP conf
1023
Jun 12 17:44:32 DEBUG[18043]: Placed channel SIP/601-d011 in ZAP conf
1023
Jun 12 17:44:32 DEBUG[18043]: Hangup: channel: -2 index = 0, normal =
41, callwait = -1, thirdcall = -1
Jun 12 17:44:32 DEBUG[18043]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-726361999
Jun 12 17:44:32 DEBUG[18043]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:32 DEBUG[18041]: Placed channel SIP/203-6116 in ZAP conf
1023
CRASH
==
==/var/log/asterisk/messages==
Jun 12 17:40:49 WARNING[17955]: No such host: 806
Jun 12 17:40:49 NOTICE[17955]: Unable to create channel of type 'SIP'
Jun 12 17:40:53 WARNING[17955]: Unable to request echo training on
channel 1
Jun 12 17:43:42 WARNING[17958]: No such host: 806
Jun 12 17:43:42 NOTICE[17958]: Unable to create channel of type 'SIP'
Jun 12 17:43:44 WARNING[17958]: Unable to request echo training on
channel 1
Jun 12 17:44:12 NOTICE[18001]: Unable to request channel SIP/595
Jun 12 17:44:12 NOTICE[18004]: Unable to request channel SIP/808
Jun 12 17:44:12 NOTICE[18008]: Unable to request channel SIP/201
Jun 12 17:44:12 NOTICE[18011]: Unable to request channel SIP/212
Jun 12 17:44:12 NOTICE[17980]: Unable to request channel SIP/704
Jun 12 17:44:12 NOTICE[17984]: Unable to request
Re: [Asterisk-Users] Which phones are good, or at least acceptable, for home and office
Steve, that happened to me too. I downloaded the public release (not beta) and it was included. I noticed that the new firmware includes a different ringer. I guess they decided we didn't need that ringer. Do you update off of their system, or do you have your own tftp server? On 6/19/06, Steve Jones [EMAIL PROTECTED] wrote: I liked the ringer that read the phone number too, but a couple months ago, I did a firmware upgrade, and that ringer option went away… Do you have the latest firmware?? I upgraded because of a problem with my phone losing registration, which is now fixed, but I lost that really cool feature… Any idea how to get that back? -Steve From: Lacy Moore - Aspendora [mailto: [EMAIL PROTECTED]] Sent: Saturday, June 17, 2006 10:21 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Which phones are good, or at least acceptable,for home and office The Grandstream seem to be a crap shoot. Some people have real good luck, others don't. So far, I've got four of them in use and the users seem to be happy. The only drawback that I have is that there is no way I can even attempt to try to explain the complex method that you have to use to PARK a call. Their attended transfers are weird. I really like the ringer that calls out the caller ID. It's because of that, that I might put them in my house. However, I still have a CIDCO device that reads out the caller ID. My house is small enough that I can hear it all over the house. I would also like to try out the Aastra 9133. It's a little more than the GXP2000. And, I have noticed the handset gets warm on the GXP. Others have mentioned this. For more information, including things already discussed about the Grandstreams, you can try: http://www.google.com/search?hl=enlr=q=site%3Ahttp%3A%2F%2Flists.digium.com%2Fpipermail%2Fasterisk-users%2F+grandstream or This site http://www.asteriskguru.com/archives/asterisk-users-vf2.html?sid=d6b13ed5fdbe515037bc9738c24f contains a complete archive of this list in forum format. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --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] Polycom Buddies in 1.6.6
All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk. Has anyone else got this to work? Doug. ___ --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] Bearer capabilities on PRI [LOOKING FOR PRI expert to resolve the issue - for hire]
Who Carez? wrote: Hey all, I am running a Asterisk 1.2.9.1 with Sangoma A101 card, newest firmware, configured with a help from Sangoma Tech Support, running fine. It is connected to a PRI circuit split from Cisco MC 3810, which in turn is connected to a Converged T from CTC Communications. While Asterisk works fine and I can call in/out on my BV account, I am only able to dial in through CTC. I have spent last 4 days researching the issue and here is what it boils down to: 1. The Asterisk box sends 809083 as bearer capabilities while the other side expects 809082. Where can I change that? Attached is the snippet from my log file. I am willing to pay to have it resolved, please email me at: responder.NOSPAM(at)pacanka(dot)com (remove .NOSPAM and replace (at) and (dot) accordingly.) Thanks, Frustrated. :) ___ --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] How to use a data T-1?
Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1. Remember, you have 1544Kbs of bandwidth. g279=8Kbs per call uLaw=64Kbs per call Just do the math. bp On 6/19/06, Warren [EMAIL PROTECTED] wrote: Steve,I want to end up with a system that will let me send and receive voicecalls.I guess what I want to do depends on the best way to do that. Can I do more than 23 (decent sounding) voice calls on a data T-1 withsomeone else handling the final part of the call to the copper for me?If so than that is my likely final destination.I have a channelized voice T-1 currently plugged into my meridian system, but I would like (if realistically possible) to do as much ofthis over IP as possible for maximum flexibility.Is that a pipe dreamor just silly given the current state of technology?I am lucky enough to work for a company that is letting me take my time with this, test the various options and come up with the propersolution.I am assuming (I know: dumb to assume) at this point thatVoIP over a T-1 to a provider that can then route it to hard phones forme would be the way to go.Similarly, I would point my 800 number to a DiD hosted by a VoIP provider that would then route the call back tome.If that is an incorrect assumption, please let me know.Regards,Warren ___ --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] DTMF Talk off
John, You said you were using a PAP2.. what is the FXO interface at the (far) asterisk end? Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.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] How to use a data T-1?
Remember to add the RTP, UDP and IP overheads. And then just do the math. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of William PiperSent: 19 June 2006 17:12To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] How to use a data T-1? Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1. Remember, you have 1544Kbs of bandwidth. g279=8Kbs per call uLaw=64Kbs per call Just do the math. bp On 6/19/06, Warren [EMAIL PROTECTED] wrote: Steve,I want to end up with a system that will let me send and receive voicecalls.I guess what I want to do depends on the best way to do that. Can I do more than 23 (decent sounding) voice calls on a data T-1 withsomeone else handling the final part of the call to the copper for me?If so than that is my likely final destination.I have a channelized voice T-1 currently plugged into my meridian system, but I would like (if realistically possible) to do as much ofthis over IP as possible for maximum flexibility.Is that a pipe dreamor just silly given the current state of technology?I am lucky enough to work for a company that is letting me take my time with this, test the various options and come up with the propersolution.I am assuming (I know: dumb to assume) at this point thatVoIP over a T-1 to a provider that can then route it to hard phones forme would be the way to go.Similarly, I would point my 800 number to a DiD hosted by a VoIP provider that would then route the call back tome.If that is an incorrect assumption, please let me know.Regards,Warren ___ --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] Transfer call via AMI or dialplan
Thanks for the tip. No idea why I missed this. Off the top of your head, does this support attended xfer, or is it a blind xfer facility ? Julian. Moises Silva wrote: Piece of cake Julian: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect Regards On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: At the moment when one of our users wants to transfer a call, they press the transfer button on the phone, enter the extension and away they go. Is there any way to do this via the AMI or dialplan ? I want them to push a button on the screen rather than using the phone itself. Julian ___ --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] Polycom Buddies in 1.6.6
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Double check to make sure you are actually running 1.6.6. I have it working with 14 extensions right now with no problems... Sean Douglas Garstang wrote: All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk. Has anyone else got this to work? Doug. ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEltC/1Kolm8VQlAURArnqAKCOTYCCwutkNjBNatQzq5yOl+XwNACguolx 0BidNydsH1rPTR1N0RZUebk= =Cvru -END PGP SIGNATURE- ___ --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] show queue ... Invalid
- Denis Shaposhnikov [EMAIL PROTECTED] wrote: What does it mean? Why is it Invalid? BTW, reload command fixes it, so the member receives queue calls. I've just reviewed the code and this should be working properly... please do a 'set debug 3' and enable the 'debug' channel in logger.conf and then try this again. You should see a message from chan_sip saying something like Checking devicestate for ... and the peername... we need to see what that message says. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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] Echo Problem with T411P
- Steve Davies [EMAIL PROTECTED] wrote: :) Now you've defeated me. I imagine that you need to do something to enable EC on that card, but it is not a card I know, so I'll leave it to someone who knows the card to offer any suggestions. The only requirement is that 'echocancel=yes' is present in zapata.conf for those channels. If the hardware echo canceler is present and enabled, then it will be used instead of the software canceler for those channels. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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 Buddies in 1.6.6
- Douglas Garstang [EMAIL PROTECTED] wrote: Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk. Has anyone else got this to work? Yes, it works on the Polycom 601 on my desk. However, the release notes say that the restriction was only removed for the IP600 and IP601; if you are using an IP300/1, IP500/1 or IP430 than the 7 buddy limit will still be in effect. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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] sangoma unicall m2rfc
Uys, Steve Underwood I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I get the far and local end unblocked but as soon as I try to make a call I get dialing and then protocol failure.. Do you guys know if there are any issues with sangoma and unicall? Anybody has an a101 card working with unicall and r2mfc? Are you out there Steve? :) ___ --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 1.07 crash under Debian Sarge
The latest version of Asterisk also includes a Page command so that
you can use that instead of an AGI script.
On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I suspect that the majority of the advice that you are going to get
would be to upgrade to the latest version of asterisk, as so many
changes and bug fixes have been made since the 1.07 release.
Julian.
Mark W. Stoddard wrote:
I have just finished implementing an Asterisk system for my place of
business (first one), and after three days of flawless usage, Asterisk
seems to have crashed. I wasn't running with '-g', so I don't have a
core dump. Here's the sequence of events leading up to the crash:
1. call comes in on our TDM2400P
2. all of our phones (about 26 Polycoms) ring. (it's after biz.
hours, so all phones ring)
3. an employee answers the call.
4. the employee attempts a page (autoanswer + meetme AGI script with
Polycoms)
5. about half the phones make it to the meeting, then the system
crashes.
6. an executive calls my manager, who's on vacation, my manager calls
me, autopsy begins.
here's a few important snippets:
===extensions.conf=
[system-page]
exten = 999,1,Macro(system-page,${CALLERIDNUM})
; The first variable is the originating caller, the others are phones I
; wish to exclude from the system-wide paging.
[macro-system-page]
exten = s,1,AGI(allpage.agi|SIP/${CALLERIDNUM});@TODO make more
robust, not only SIP
exten = s,2,MeetMe(999,Adqt)
;exten = s,2,Hangup
[add-to-page]
exten = listener,1,MeetMe(999,dmqx)
===
==/var/log/asterisk/debug==
Jun 12 17:44:12 DEBUG[17975]: Building dynamic conference '999'
Jun 12 17:44:12 DEBUG[17975]: Placed channel SIP/302-6188 in ZAP conf
1023
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
...
Jun 12 17:44:18 DEBUG[17975]: Hangup: channel: -2 index = 0, normal =
51, callwait = -1, thirdcall = -1
Jun 12 17:44:18 DEBUG[17975]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-1321090091
Jun 12 17:44:18 DEBUG[17975]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:19 DEBUG[17975]: update_user_counter(302) - decrement inUse
counter
Jun 12 17:44:19 DEBUG[18016]: Building dynamic conference '999'
Jun 12 17:44:20 DEBUG[18016]: Placed channel SIP/508-af01 in ZAP conf
1023
Jun 12 17:44:20 DEBUG[18016]: Hangup: channel: -2 index = 0, normal =
41, callwait = -1, thirdcall = -1
Jun 12 17:44:20 DEBUG[18016]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-1583015986
Jun 12 17:44:20 DEBUG[18016]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:21 DEBUG[18016]: update_user_counter(508) - decrement
outUse counter
Jun 12 17:44:21 DEBUG[23992]: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 103: Found
Jun 12 17:44:21 DEBUG[18017]: Building dynamic conference '999'
Jun 12 17:44:22 DEBUG[18017]: Placed channel SIP/804-677b in ZAP conf
1023
Jun 12 17:44:22 DEBUG[18017]: Hangup: channel: -2 index = 0, normal =
41, callwait = -1, thirdcall = -1
Jun 12 17:44:22 DEBUG[18017]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-1132503448
Jun 12 17:44:22 DEBUG[18017]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:23 DEBUG[18017]: update_user_counter(804) - decrement
outUse counter
...
Jun 12 17:44:32 DEBUG[18041]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18019]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18021]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18028]: update_user_counter(404) - decrement
outUse counter
Jun 12 17:44:32 DEBUG[18042]: Placed channel SIP/401-1bec in ZAP conf
1023
Jun 12 17:44:32 DEBUG[18043]: Placed channel SIP/601-d011 in ZAP conf
1023
Jun 12 17:44:32 DEBUG[18043]: Hangup: channel: -2 index = 0, normal =
41, callwait = -1, thirdcall = -1
Jun 12 17:44:32 DEBUG[18043]: Set option TDD MODE, value: OFF(0) on
Zap/pseudo-726361999
Jun 12 17:44:32 DEBUG[18043]: Updated conferencing on -2, with 0
conference users
Jun 12 17:44:32 DEBUG[18041]: Placed channel SIP/203-6116 in ZAP conf
1023
CRASH
==
==/var/log/asterisk/messages==
Jun 12 17:40:49 WARNING[17955]: No such host: 806
Jun 12 17:40:49 NOTICE[17955]: Unable to create channel of type 'SIP'
Jun 12 17:40:53 WARNING[17955]: Unable to request echo training on
channel 1
Jun 12 17:43:42 WARNING[17958]: No such host: 806
Jun 12 17:43:42 NOTICE[17958]: Unable to create channel of type 'SIP'
Jun 12 17:43:44 WARNING[17958]: Unable to request echo training on
channel 1
Jun 12 17:44:12 NOTICE[18001]: Unable to request channel SIP/595
Jun 12 17:44:12 NOTICE[18004]: Unable to request channel
[Asterisk-Users] Linksys PAP2NA Configuration / Asterisk / Voip consultant wanted
Anyone on the list good with Linksys PAP2NA configuration, I am looking to take my atas and emulate the operation of a pots phone line as close as I can get. One thing I need to change is the fast busy tone I get when someone hangs up on the call. We are also looking for a Voip/ Asterisk Consultant to set up hardware for a call center application. We plan to use an autodialer based on analog dialogic boards to interface with a 16- 24 port analog voip gateway. The goal is to make the gateway act just like normal pots lines with regards to disconnects and what tones are played. Mark Adams Email me off list We are willing to pay for effective results [EMAIL PROTECTED] ___ --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] sip to h323 ... direct RTP?
Hi, Thanks to those who hinted on the SIP/H323/Skinny capabilities of asterisk ... I am starting to like this app! :D Now, I successfully managed to bridge SIP to H323 (i don't have skinny phones here). Just a question: Is it possible to have Asterisk just as a signalling proxy? i have a flat test network, and i would like the RTP streams to be sent directly end to end (sip phone to h323 phone). It should be possible ... but is it possible with asterisk? Thanks! Cesc ___ --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] Custom extension halting execution upon caller hanging up
Hello, list!
I'm having some trouble with [EMAIL PROTECTED] 2.7(?), Asterisk 1.2.5, inasmuch as
my custom extension is not continuing execution when the caller hangs
up. (Please excuse the sterilized output.)
Here's how it's supposed to go:
exten = 2,8,Monitor(wav,${TIMESTAMP})
exten = 2,9,Dial(SIP/Provider/8005551212)
exten = 2,10,Macro(record-cleanup)
If the caller hangs up before the callee does, execution of the
custom extension halts and does not continue to priority 10
(record-cleanup), where sox is used to reverse the audio files and
then mix them then reverse them again so they'll be in sync (since
inbound audio only starts from call-answered but outbound audio
starts from the beginning of ringback).
Asterisk provides this debug output to the console (internal
extension 101 is the caller):
-- Called Provider/8005551212
-- SIP/Provider-993d is making progress passing it to SIP/101-1666
-- SIP/Provider-993d answered SIP/101-1666
The call proceeds normally, but then Asterisk spits this out the
moment the caller hangs up first:
== Spawn extension (custom-extension, 2, 9) exited non-zero on
'SIP/101-1666'
How can I prevent the extension from bailing before I have a chance
to clean up the recording in priority 10?
Thanks in advance!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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[Asterisk-Users] Question about context from-internal
All: I tested echo test by dialing *43 under Asterisk configured by FreePbx by using x-lite softphone. I could not figure out how the call is routed to context from-internal. In sip_additional.conf, I have three extensions defined as 2826, 2800 and 2801, which all are defined context as from-internal. FreePbx doesn't define any entry for *43 as an extension in sip related config files. I greped from-internal for all configure files, I did not find any definition of context from-internal in sip related files, except three sip extensions. After I dialed *43, cdr did show the content of dcontext as from-internal. Can anyone explain how the call to reach the context from-internal, or how do I get a context trace for a call? Many thanks. Tielin ___ --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] Act-Tel G11112DS Telephony Gateway
Hey everyone, I recently bought an Act-Tel G2DS telephony gateway (the web interface says it's model # is GS though.) Has anyone else on this list used one of these? It has one FXO and one FXS port. I have an account for it set up in sip.conf on my Asterisk box and it apparently logs in correctly because I can dial the extension I set up in extensions.conf and the FXS port rings and I can answer it. However, I cannot dial out through my Asterisk box on it. I need to get this part working before I even think of trying to put my dial tone on the FXO port. So, has anyone use one of these and might have some kind of documentation for it? Thanks Undrhil ___ --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] finding mac addresses
All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools.Thanks-John ___ --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] How to use a data T-1?
So let's assume I am going to use G.729A. I am looking at using Polycom IP601 phones which support G729A directly, so the only licenses I believe I would need are for the calls going to voicemail or in the menu system at once - realistically that number never exceeds 5 simultaneous, since the phones can handle the CODEC and no transcoding is needed, so those do not need licenses according to http://www.voip-info.org/wiki-Asterisk+G.729+Licensing. It looks to me like, for testing, I can get a couple of the polycom phones and have a server using an IP on the unused T1. Assuming that is correct (which I will write up as an article for the Wiki if anyone is interested when this is all done), the next thing I need is a provider of VoIP service. Also, it seems like the server would go on the outside of my firewall with holes punched through for the phones which would be on the ind=side of the firewall. Would that be correct? W Steve Langstaff wrote: Remember to add the RTP, UDP and IP overheads. And then just do the math. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of William Piper Sent: 19 June 2006 17:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1. Remember, you have 1544Kbs of bandwidth. g279=8Kbs per call uLaw=64Kbs per call Just do the math. bp On 6/19/06, Warren [EMAIL PROTECTED] wrote: Steve, I want to end up with a system that will let me send and receive voice calls.I guess what I want to do depends on the best way to do that. Can I do more than 23 (decent sounding) voice calls on a data T-1 with someone else handling the final part of the call to the copper for me? If so than that is my likely final destination. I have a channelized voice T-1 currently plugged into my meridian system, but I would like (if realistically possible) to do as much of this over IP as possible for maximum flexibility.Is that a pipe dream or just silly given the current state of technology? I am lucky enough to work for a company that is letting me take my time with this, test the various options and come up with the proper solution.I am assuming (I know: dumb to assume) at this point that VoIP over a T-1 to a provider that can then route it to hard phones for me would be the way to go.Similarly, I would point my 800 number to a DiD hosted by a VoIP provider that would then route the call back to me.If that is an incorrect assumption, please let me know. Regards, Warren ___ --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] Re: Two FXO: How to dial a number when a RING comes in?
Thanks Noah for the help, but... no go :-/ From: Noah Miller ONE: You should answer an incoming zap line before doing anything with it, so do this: exten = s,1,Answer exten = s,2,Dial(Zap/2/014XX) When I try this, instead of using the Zap/2 interface to ring the other number, Asterisk goes off hook and I hear some kind of static: Jun 19 18:17:46 NOTICE[2186] chan_zap.c: Got event 18 (Ring Begin)... Jun 19 18:17:47 NOTICE[2186] chan_zap.c: Got event 2 (Ring/Answered)... Jun 19 18:17:51 NOTICE[2186] chan_zap.c: Got event 18 (Ring Begin)... TWO: Are there any console messages? Can you dial into the system and get internal extensions? Maybe you could try a testing dialplan like this: exten = s,1,Answer exten = s,2,Waitexten(10) exten = 100,Dial(Zap/2/014XX) Then call in and after you're connected, dial 100 to see if it will dial out on ZAP/2 When I try this, /var/log/asterisk/messages says: Jun 19 18:12:38 NOTICE[1660] pbx.c: Cannot find extension '100' in context '(null)' Jun 19 18:12:38 WARNING[1660] pbx_config.c: Invalid priority/label 'Dial' at line 172 I just realized that I blindly typed the above, without realizing that the second parameter is missing. Regardless, since even the first test doesn't work... Just in case, I'd like to repeat that I don't want Asterisk to answer the call: I just want it to use the second FXO to ring another phone, at a remote location. For reference, I went back to the original configuration that I used, but it picks up the line and remains silent (static noises): --- extensions.conf -- [cherbourg] exten = s,1,Dial(Zap/2/0145815059) --- zaptel.conf --- fxsks=1,2 loadzone=fr defaultzone=fr zapata.conf --- [channels] ;context=default context=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid=my caller id(123) 123-1234 channel=1 ;context=default context=cherbourg signalling=fxs_ks usecallerid=yes echocancel=yes callgroup=1 pickupgroup=1 immediate=no callerid=my caller id(123) 123-1234 channel=2 and just in case you're wondering if the FXO cards are correctly loaded... - dmesg - Jun 19 18:12:31 localhost syslogd 1.4.1: restart. Jun 19 18:12:31 localhost kernel: klogd 1.4.1, log source = /proc/kmsg started. Jun 19 18:12:31 localhost kernel: Linux version 2.6.13.4-1.x86.i686.cmov ([EMAIL PROTECTED]:1) (gcc version 3.4.4) #1 Wed Nov 23 11:31:48 EST 2005 [...] Jun 19 18:12:31 localhost kernel: Zapata Telephony Interface Registered on major 196 Jun 19 18:12:31 localhost kernel: Zaptel Version: Echo Canceller: KB1 Jun 19 18:12:31 localhost kernel: Registered Tormenta2 PCI Jun 19 18:12:31 localhost kernel: ACPI: PCI Interrupt Link [LNKA] enabled at IRQ 5 Jun 19 18:12:31 localhost kernel: PCI: setting IRQ 5 as level-triggered Jun 19 18:12:31 localhost kernel: ACPI: PCI Interrupt :00:08.0[A] - Link [LNKA] - GSI 5 (level, low) - IRQ 5 Jun 19 18:12:32 localhost kernel: wcfxo: DAA mode is 'FCC' Jun 19 18:12:32 localhost kernel: Found a Wildcard FXO: Generic Clone Jun 19 18:12:32 localhost kernel: ACPI: PCI Interrupt Link [LNKD] enabled at IRQ 10 Jun 19 18:12:32 localhost kernel: PCI: setting IRQ 10 as level-triggered Jun 19 18:12:32 localhost kernel: ACPI: PCI Interrupt :00:09.0[A] - Link [LNKD] - GSI 10 (level, low) - IRQ 10 Jun 19 18:12:32 localhost kernel: wcfxo: DAA mode is 'FCC' Jun 19 18:12:32 localhost kernel: Found a Wildcard FXO: Generic Clone Jun 19 18:12:32 localhost kernel: usbcore: registered new driver wcusb Jun 19 18:12:32 localhost kernel: Wildcard USB FXS Interface driver registered Jun 19 18:12:35 localhost kernel: Registered tone zone 2 (France) = Surely, I can't be the only one in this list who needs to set up Asterisk simply to ring a remote phone when a call comes in at the office. Anybody has a working configuration that I could use as a reference? Thank you :-) VD. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.0/368 - Release Date: 16/06/2006 ___ --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] finding mac addresses
arp in the shell mojowrkn wrote: All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools. Thanks-John ___ --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] finding mac addresses
as long as they are in the same network segment as the asterisk server you can use arp man arp mojowrkn wrote: All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools. Thanks-John ___ --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] How to use a data T-1?
After all the overhead, for uLaw you would need about 90kbps (give or take) and for G.729, you would need about 32kbps (give or take). Therefore, you would have the following: uLaw= about 17 calls g729= about 48 calls I am trying to start a voip service in my local area and sometimes seeing these numbers make me wonder how using VoIP for larger companies could possibly be profitable if you require a $500+ data T1 just have a decent connect (unless you use g729?) - Gabe Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1. Remember, you have 1544Kbs of bandwidth. g279=8Kbs per call uLaw=64Kbs per call Just do the math. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Talk off
Doug,
The interface that i dial to is at my Service provider and am not sure what
they are using. I dial out of my * box to a service provider number which is
answerd by an asterisk box that I have at another location on a high speed
cable connection, that box then dials the numberI ultimately want to reach.
I use an extensions.conf line at my home * such as:
Dial(zap/1/my_sip_numberww${EXTEN});
this works great and saves me a ton on call costs.
John
On Monday June 19 2006 12:19 pm, Doug Crompton wrote:
John,
You said you were using a PAP2.. what is the FXO interface at the (far)
asterisk end?
Doug
* Doug Crompton *
* Richboro, PA 18954*
* 215-431-6307 *
**
* [EMAIL PROTECTED]*
* http://www.crompton.com *
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Re: [Asterisk-Users] finding mac addresses
[EMAIL PROTECTED] root]# arp -an ? (172.16.8.1) at 00:04:C1:21:CC:C0 [ether] on eth0 ? (172.16.8.53) at 00:04:F2:01:FA:94 [ether] on eth0 ? (172.16.8.48) at 00:04:F2:01:FA:D8 [ether] on eth0 ? (172.16.8.62) at 00:04:F2:01:FB:65 [ether] on eth0 ? (172.16.8.60) at 00:04:F2:01:FB:20 [ether] on eth0 ? (172.16.8.59) at 00:07:E9:50:F7:56 [ether] on eth0 ? (172.16.8.58) at 00:04:F2:00:D0:23 [ether] on eth0 ? (172.16.8.57) at 00:07:E9:50:F7:4D [ether] on eth0 mojowrkn wrote: All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Echo Problem with T411P
Kevin P. Fleming wrote: - Steve Davies [EMAIL PROTECTED] wrote: :) Now you've defeated me. I imagine that you need to do something to enable EC on that card, but it is not a card I know, so I'll leave it to someone who knows the card to offer any suggestions. The only requirement is that 'echocancel=yes' is present in zapata.conf for those channels. If the hardware echo canceler is present and enabled, then it will be used instead of the software canceler for those channels. How can you detect if the HW echo can is enabled? Is it console output during module load or something else? ___ --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] finding mac addresses
mojowrkn wrote: All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools. Thanks-John ___ --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 arp -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] finding mac addresses
From your Asterisk console: tcpdump -i eth0 -e | grep -A1 your target phone's IP address Then: Make a call on your target phone. Disclaimer: not tested -Original Message-From: mojowrkn [mailto:[EMAIL PROTECTED]Sent: Monday, June 19, 2006 11:21 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] finding mac addressesAll, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools.Thanks-John ___ --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] finding mac addresses
On Mon, Jun 19, 2006 at 10:21:16AM -0700, mojowrkn wrote: All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools. Is it in your LAN? if so, arp(8) is your friend. /proc/net/arp likewise. You may need to ping the phone beforehand to make sure its address is actually in the table. If the phone is not in your LAN, there is no direct way to tell. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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] finding mac addresses
If they are on the same network you can do the following:
arp -a | grep $IPADDRESS |awk '{print $4}'
you may need to adjust awk(ed) position due to you distro.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Casey Boone
Sent: Monday, June 19, 2006 1:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] finding mac addresses
as long as they are in the same network segment as the asterisk server
you can use arp
man arp
mojowrkn wrote:
All, Can anyone point me to the best way to find the mac address of
a
phone on my system?? I can get the ip's just fine but dont seem to
be
able to pull mac addresses from any network tools.
Thanks-John
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Re: [Asterisk-Users] Bristuff-0.3.0-PRE-1q and florz patch compile trouble
I cannot help you with the problem, I can only tell you it works for me (on a Debian system) I wonder what the florz patch is though. I never used it, but I hear some ppl about it all the time. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] finding mac addresses
an example might be
IP=10.0.0.213
MAC=`arp | grep $IP | awk {'print $3'}`
mojowrkn wrote:
All, Can anyone point me to the best way to find the mac address of a
phone on my system?? I can get the ip's just fine but dont seem to be
able to pull mac addresses from any network tools.
Thanks-John
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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] finding mac addresses
If your phones are connected to a Cisco switch, depending your your IOS level you can possibly use the show mac-address-table command. Which would show you not only the mac-address for all the devices attached to the switch, but what port they are hanging off of. Hope this helps. T. From: mojowrkn [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] finding mac addresses Date: Mon, 19 Jun 2006 10:21:16 -0700 All, Can anyone point me to the best way to find the mac address of a phone on my system?? I can get the ip's just fine but dont seem to be able to pull mac addresses from any network tools. Thanks-John ___ --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] How to use a data T-1?
If your T1 is currently configured for connecting you to the Internet, then your Asterisk just becomes a client on your network, and can terminate calls to Internet based providers by SIP or IAX. No reason for a T1 card or connection to the Asterisk. I don't have enough experience to say who may be the most reliable provider, but you can use any of them for testing. Others have given details of bandwidth requirements for the different codecs, and know more than I about that.. Once you get the basics connected, then any 800# provider should be able to point a number to any existing DID, or you can use a VoIP provider to provide an 800# directly. -Steve -Original Message- From: Warren [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? Steve, I want to end up with a system that will let me send and receive voice calls. I guess what I want to do depends on the best way to do that. Can I do more than 23 (decent sounding) voice calls on a data T-1 with someone else handling the final part of the call to the copper for me? If so than that is my likely final destination. I have a channelized voice T-1 currently plugged into my meridian system, but I would like (if realistically possible) to do as much of this over IP as possible for maximum flexibility. Is that a pipe dream or just silly given the current state of technology? I am lucky enough to work for a company that is letting me take my time with this, test the various options and come up with the proper solution. I am assuming (I know: dumb to assume) at this point that VoIP over a T-1 to a provider that can then route it to hard phones for me would be the way to go. Similarly, I would point my 800 number to a DiD hosted by a VoIP provider that would then route the call back to me. If that is an incorrect assumption, please let me know. Regards, Warren Steve Jones wrote: Depends what you want to do! Do you want to do VoIP over that T1 to a provider or IP telephones? Do you want to hook up to the PSTN through that T1 as 24 voice channels, through a T1 card on your asterisk? If you want to use the T1 as 24 voice channels, the Telco is going to have to re-provision the T1 as a voice T1, because currently, presumably it is one big channel of data. You could have the telco do any combination of 24 channels, some voice and some data, if your DSU or router allows drop and insert of channels. It would then split the T1 into a voice side and a data side, each with part of the channels available. Once you have a channelized voice T1, it can plug into a voice T1 card in your Asterisk, but typically can't do data anymore, so if that's not what you intend, then please explain further.. -Original Message- From: Warren [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How to use a data T-1? I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren ___ --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] Which phones are good, or at least acceptable, for home and office
I found a message on this list, that provided a recommendation to use 195.140.132.34, which I think is a non-afflilated someone that just happens to be providing tested firmwares. I couldn't get the default to work... What server do you use? What firmware do you have? I've got a GS100... Here's my info: Product Model: BT100 REV 2.0 Software Version: Program-- 1.0.8.16 Bootloader-- 1.0.8.9 HTML-- 1.0.8.16 VOC-- 1.0.1.0 From: Lacy Moore - Aspendora [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 11:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Which phones are good, or at least acceptable,for home and office Steve, that happened to me too. I downloaded the public release (not beta) and it was included. I noticed that the new firmware includes a different ringer. I guess they decided we didn't need that ringer. Do you update off of their system, or do you have your own tftp server? On 6/19/06, Steve Jones [EMAIL PROTECTED] wrote: I liked the ringer that read the phone number too, but a couple months ago, I did a firmware upgrade, and that ringer option went away... Do you have the latest firmware?? I upgraded because of a problem with my phone losing registration, which is now fixed, but I lost that really cool feature... Any idea how to get that back? -Steve From: Lacy Moore - Aspendora [mailto: [EMAIL PROTECTED] Sent: Saturday, June 17, 2006 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Which phones are good, or at least acceptable,for home and office The Grandstream seem to be a crap shoot. Some people have real good luck, others don't. So far, I've got four of them in use and the users seem to be happy. The only drawback that I have is that there is no way I can even attempt to try to explain the complex method that you have to use to PARK a call. Their attended transfers are weird. I really like the ringer that calls out the caller ID. It's because of that, that I might put them in my house. However, I still have a CIDCO device that reads out the caller ID. My house is small enough that I can hear it all over the house. I would also like to try out the Aastra 9133. It's a little more than the GXP2000. And, I have noticed the handset gets warm on the GXP. Others have mentioned this. For more information, including things already discussed about the Grandstreams, you can try: http://www.google.com/search?hl=enlr=q=site%3Ahttp%3A%2F%2Flists.digium.com%2Fpipermail%2Fasterisk-users%2F+grandstream or This site http://www.asteriskguru.com/archives/asterisk-users-vf2.html?sid=d6b13ed5fdbe515037bc9738c24f contains a complete archive of this list in forum format. ___ --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 -- Lacy Moore Aspendora, Inc. ___ --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] How to use a data T-1?
I honestly do not see the big deal about using g729. It is a one-time fee and you would only need to buy as many licenses as you have people in ivr or voicemail if you have g729 phones. For a business this is not a major expense. You are talking about spending $100-$200 (max $480 for all 48 potential callers if you don't have g729 phones) to expand a T-1 from 23 calls (PRI) to 48 calls by your measurement - a doubling of the usage of the T-1 for less than one month's cost of the T-1. ROI at less than a month? That's a slam-dunk for most businesses. W Gabriel Afana wrote: After all the overhead, for uLaw you would need about 90kbps (give or take) and for G.729, you would need about 32kbps (give or take). Therefore, you would have the following: uLaw= about 17 calls g729= about 48 calls I am trying to start a voip service in my local area and sometimes seeing these numbers make me wonder how using VoIP for larger companies could possibly be profitable if you require a $500+ data T1 just have a decent connect (unless you use g729?) - Gabe ___ --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] How to use a data T-1?
If youre going to have to open ports on your firewall for SIP anyway, then why not put the server on the inside? That being said, I dont know if youd need to punch holes for the phones being trusted and the server on the outside.. Personally I dont like the ideas of having a server outside, but maybe Im too paranoid?! From: Warren [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? So let's assume I am going to use G.729A. I am looking at using Polycom IP601 phones which support G729A directly, so the only licenses I believe I would need are for the calls going to voicemail or in the menu system at once - realistically that number never exceeds 5 simultaneous, since the phones can handle the CODEC and no transcoding is needed, so those do not need licenses according to http://www.voip-info.org/wiki-Asterisk+G.729+Licensing. It looks to me like, for testing, I can get a couple of the polycom phones and have a server using an IP on the unused T1. Assuming that is correct (which I will write up as an article for the Wiki if anyone is interested when this is all done), the next thing I need is a provider of VoIP service. Also, it seems like the server would go on the outside of my firewall with holes punched through for the phones which would be on the ind=side of the firewall. Would that be correct? W Steve Langstaff wrote: Remember to add the RTP, UDP and IP overheads. And then just do the math. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of William Piper Sent: 19 June 2006 17:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1. Remember, you have 1544Kbs of bandwidth. g279=8Kbs per call uLaw=64Kbs per call Just do the math. bp On 6/19/06, Warren [EMAIL PROTECTED] wrote: Steve, I want to end up with a system that will let me send and receive voice calls.I guess what I want to do depends on the best way to do that. Can I do more than 23 (decent sounding) voice calls on a data T-1 with someone else handling the final part of the call to the copper for me? If so than that is my likely final destination. I have a channelized voice T-1 currently plugged into my meridian system, but I would like (if realistically possible) to do as much of this over IP as possible for maximum flexibility.Is that a pipe dream or just silly given the current state of technology? I am lucky enough to work for a company that is letting me take my time with this, test the various options and come up with the proper solution.I am assuming (I know: dumb to assume) at this point that VoIP over a T-1 to a provider that can then route it to hard phones for me would be the way to go.Similarly, I would point my 800 number to a DiD hosted by a VoIP provider that would then route the call back to me.If that is an incorrect assumption, please let me know. Regards, Warren ___--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] How to use a data T-1?
I would say its only profitable if youre getting ONE T1 instead of two?? From: Gabriel Afana [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 1:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? After all the overhead, for uLaw you would need about 90kbps (give or take) and for G.729, you would need about 32kbps (give or take). Therefore, you would have the following: uLaw= about 17 calls g729= about 48 calls I am trying to start a voip service in my local area and sometimes seeing these numbers make me wonder how using VoIP for larger companies could possibly be profitable if you require a $500+ data T1 just have a decent connect (unless you use g729?) - Gabe Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1. Remember, you have 1544Kbs of bandwidth. g279=8Kbs per call uLaw=64Kbs per call Just do the math. ___ --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 enter an extension to dial while voicemail is playing?
I have a very, very simple Asterisk setup in my house. I have a Sipura 3000 with a PSTN line connected and one analog phone connected. The [incoming] context looks like this: exten = s,1,Dial(SIP/50,23,r) exten = s,2,VoiceMail([EMAIL PROTECTED]) exten = s,3,Playback(vm-goodbye) exten = s,4,Hangup As you can see, when somebody calls in if I don't answer in 23 seconds then they are forwarded to my voicemail. How can I make it so I can call an enter extensions either while the phone is ringing or while the voicemail message is playing? I want the system to be as seemless as possible so the wife is happy =) Right now it works great because my Sipura 3000 forwards to call to Asterisk and Asterisk rings my analog phone, but the incoming caller hears a steady dial-tone the whole time. I wouldn't want that to change. (so the caller isn't wondering what is going on) Any help is appriciated :) ___ --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] Echo Problem with T411P
On 6/19/06, Mike Fedyk [EMAIL PROTECTED] wrote: Kevin P. Fleming wrote: - Steve Davies [EMAIL PROTECTED] wrote: :) Now you've defeated me. I imagine that you need to do something to enable EC on that card, but it is not a card I know, so I'll leave it to someone who knows the card to offer any suggestions. The only requirement is that 'echocancel=yes' is present in zapata.conf for those channels. If the hardware echo canceler is present and enabled, then it will be used instead of the software canceler for those channels. How can you detect if the HW echo can is enabled? Is it console output during module load or something else? Yes. You'll see messages about a VPM (Voice Processing Module) getting initialized. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] finding mac addresses
On Mon, Jun 19, 2006 at 01:52:39PM -0400, Alexander Lopez wrote:
If they are on the same network you can do the following:
arp -a | grep $IPADDRESS |awk '{print $4}'
grep before awk?
arp -n | awk '/^IPADDRESS / {print $3}'
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755 iax:[EMAIL PROTECTED]
+972-50-7952406
[EMAIL PROTECTED] http://www.xorcom.com
___
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Re: [Asterisk-Users] Bristuff-0.3.0-PRE-1q and florz patch compile trouble
found it, in bristuff-0.3.0-PRE-1q/zaphfc/Makefile again it is required to change KSRC=/usr/src/linux/ to KSRC=/usr/src/linux-2.6/ I wonder why neither florz nor kapejod fixes these problems (several modules do not compile). I will not try running bristuff anymore without florz but from the time when I was running asterisk-1.0.9 i got crashes, lost bri lines and about every 3 days a complete lockup of the box without florz patch. But I guess if bristuff is running without problems for you probably it is not needed for your setup. It should solve all sorts of timing problems (Or mmaybe it's because I'm running CentOS x86_64, for a production asterisk box I just used i386 CentOS) On Mon, 19 Jun 2006, Michiel van Baak wrote: I cannot help you with the problem, I can only tell you it works for me (on a Debian system) I wonder what the florz patch is though. I never used it, but I hear some ppl about it all the time. ___ --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: DTMF Talk off
With recent versions of *, you can increase the detection time in zapata.conf with the toneduration variable. The default setting is: toneduration=100 We had the same problem and solved it by increasing this to 200. You can also increase the threshold volume for detection of DTMF by setting VPM_DEFAULT_DTMFTHRESHOLD in the relevant zaptel wctX.c and recompiling (though if you increase this too much you risk losing your ability to detect DTMF at all). Hope this helps, Matt. ___ --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 Buddies in 1.6.6
I am running an IP601 on my desk and it is only monitoring up to 8. If I add more, it drops the oldest and adds the new one. running 1.6.6.0036 On Jun 19, 2006, at 11:40 AM, Kevin P. Fleming wrote: - Douglas Garstang [EMAIL PROTECTED] wrote: Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk. Has anyone else got this to work? Yes, it works on the Polycom 601 on my desk. However, the release notes say that the restriction was only removed for the IP600 and IP601; if you are using an IP300/1, IP500/1 or IP430 than the 7 buddy limit will still be in effect. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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] How to use a data T-1?
Well that is certainly all good news. The last hardware question I would then have is: What do you do for Echo Cancellation with this type of setup? Everyone keeps saying that the software EC basically sucks to put it bluntly. Is there some sort of hardware to do EC that can be used here? W Steve Jones wrote: If your T1 is currently configured for connecting you to the Internet, then your Asterisk just becomes a client on your network, and can terminate calls to Internet based providers by SIP or IAX. No reason for a T1 card or connection to the Asterisk. I don't have enough experience to say who may be the most reliable provider, but you can use any of them for testing. Others have given details of bandwidth requirements for the different codecs, and know more than I about that.. Once you get the basics connected, then any 800# provider should be able to point a number to any existing DID, or you can use a VoIP provider to provide an 800# directly. -Steve ___ --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] sip to h323 ... direct RTP?
19 jun 2006 kl. 19.02 skrev Cesc: Hi, Thanks to those who hinted on the SIP/H323/Skinny capabilities of asterisk ... I am starting to like this app! :D Now, I successfully managed to bridge SIP to H323 (i don't have skinny phones here). Just a question: Is it possible to have Asterisk just as a signalling proxy? i have a flat test network, and i would like the RTP streams to be sent directly end to end (sip phone to h323 phone). It should be possible ... but is it possible with asterisk? No. It's certainly possible but at this time there's no interaction between the RTP clients, the various channel drivers. /Olle --- Olle E. Johansson * Asterisk Evangelist, developer * VOOP A/S [EMAIL PROTECTED] ___ --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: Can I enter an extension to dial while voicemail is playing?
Using the Background command, you will be able to play the voicemail while still being allowed to enter digits. exten = s,1,Wait(2) exten = 108,2,Background(voicemail/default/108/unavail) exten = s,1,Dial(SIP/50,23,r) exten = s,2,Background(/voicemail/default/50/unavail) ;or whatever the soundfile is called exten = s,3,Voicemail(s50) ;s will skip the greeting and just go to the beep exten = s,4,Playback(vm-goodbye) exten = s,5,Hangup You can then put exten = 1, Dial(sip/me) exten = 2, Dial(sip/her) or whatever your dial statements look like. Leah Newmark Capalon VoIP [EMAIL PROTECTED] wrote: Message: 9 Date: Mon, 19 Jun 2006 14:18:22 -0400 From: John Klimek [EMAIL PROTECTED] Subject: [Asterisk-Users] Can I enter an extension to dial while voicemail is playing? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I have a very, very simple Asterisk setup in my house. I have a Sipura 3000 with a PSTN line connected and one analog phone connected. The [incoming] context looks like this: exten = s,1,Dial(SIP/50,23,r) exten = s,2,VoiceMail([EMAIL PROTECTED]) exten = s,3,Playback(vm-goodbye) exten = s,4,Hangup As you can see, when somebody calls in if I don't answer in 23 seconds then they are forwarded to my voicemail. How can I make it so I can call an enter extensions either while the phone is ringing or while the voicemail message is playing? I want the system to be as seemless as possible so the wife is happy =) Right now it works great because my Sipura 3000 forwards to call to Asterisk and Asterisk rings my analog phone, but the incoming caller hears a steady dial-tone the whole time. I wouldn't want that to change. (so the caller isn't wondering what is going on) Any help is appriciated :) ___ --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] home routers
I'm looking for somehting like the standard house hold linksys/dlink router. Basically it needs to have at least 1x100mbit port, wireless G capabilitys and at least 1 x anolog voip/sip connection. I've found linksys's WRT54GP2 which appears to do what i want. Anybody use this? Does it require vontage's service? I'm looking for any recommendations. Thanks -- ~Shaun ___ --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 1.07 crash under Debian Sarge
True, there have been many fixes since then. I would at least consider
upgrading Asterisk+zaptel to the latest 1.0x which I think is 1.09.
If you want to try troubleshoot it first I would watch my memory usage over
the next couple days for memory leaks. If you find your using more and more
physical memory over time and don't want to upgrade I would schedule a cron
job to reboot the system every night. It's not a pretty solution but it
get's the job done.
What sort of hardware platform are you using? Adequate cooling? Clean
power?
I would definitely be interested to know what you find.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Monday, June 19, 2006 8:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.07 crash under Debian Sarge
I suspect that the majority of the advice that you are going
to get would be to upgrade to the latest version of asterisk,
as so many changes and bug fixes have been made since the
1.07 release.
Julian.
Mark W. Stoddard wrote:
I have just finished implementing an Asterisk system for my
place of
business (first one), and after three days of flawless
usage, Asterisk
seems to have crashed. I wasn't running with '-g', so I
don't have a
core dump. Here's the sequence of events leading up to the crash:
1. call comes in on our TDM2400P
2. all of our phones (about 26 Polycoms) ring. (it's after biz.
hours, so all phones ring)
3. an employee answers the call.
4. the employee attempts a page (autoanswer + meetme AGI
script with
Polycoms)
5. about half the phones make it to the meeting, then the system
crashes.
6. an executive calls my manager, who's on vacation, my
manager calls
me, autopsy begins.
here's a few important snippets:
===extensions.conf=
[system-page]
exten = 999,1,Macro(system-page,${CALLERIDNUM})
; The first variable is the originating caller, the others
are phones
I ; wish to exclude from the system-wide paging.
[macro-system-page]
exten = s,1,AGI(allpage.agi|SIP/${CALLERIDNUM})
;@TODO make more
robust, not only SIP
exten = s,2,MeetMe(999,Adqt)
;exten = s,2,Hangup
[add-to-page]
exten = listener,1,MeetMe(999,dmqx)
===
==/var/log/asterisk/debug==
Jun 12 17:44:12 DEBUG[17975]: Building dynamic conference '999'
Jun 12 17:44:12 DEBUG[17975]: Placed channel SIP/302-6188
in ZAP conf
1023
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
Jun 12 17:44:12 DEBUG[17979]: Manager received command 'Originate'
...
Jun 12 17:44:18 DEBUG[17975]: Hangup: channel: -2 index =
0, normal =
51, callwait = -1, thirdcall = -1 Jun 12 17:44:18 DEBUG[17975]: Set
option TDD MODE, value: OFF(0) on
Zap/pseudo-1321090091
Jun 12 17:44:18 DEBUG[17975]: Updated conferencing on -2, with 0
conference users Jun 12 17:44:19 DEBUG[17975]:
update_user_counter(302) - decrement inUse counter Jun 12 17:44:19
DEBUG[18016]: Building dynamic conference '999'
Jun 12 17:44:20 DEBUG[18016]: Placed channel SIP/508-af01
in ZAP conf
1023
Jun 12 17:44:20 DEBUG[18016]: Hangup: channel: -2 index =
0, normal =
41, callwait = -1, thirdcall = -1 Jun 12 17:44:20 DEBUG[18016]: Set
option TDD MODE, value: OFF(0) on
Zap/pseudo-1583015986
Jun 12 17:44:20 DEBUG[18016]: Updated conferencing on -2, with 0
conference users Jun 12 17:44:21 DEBUG[18016]:
update_user_counter(508) - decrement outUse counter Jun 12 17:44:21
DEBUG[23992]: Stopping retransmission on
'[EMAIL PROTECTED]' of Request
103: Found
Jun 12 17:44:21 DEBUG[18017]: Building dynamic conference '999'
Jun 12 17:44:22 DEBUG[18017]: Placed channel SIP/804-677b
in ZAP conf
1023
Jun 12 17:44:22 DEBUG[18017]: Hangup: channel: -2 index =
0, normal =
41, callwait = -1, thirdcall = -1 Jun 12 17:44:22 DEBUG[18017]: Set
option TDD MODE, value: OFF(0) on
Zap/pseudo-1132503448
Jun 12 17:44:22 DEBUG[18017]: Updated conferencing on -2, with 0
conference users Jun 12 17:44:23 DEBUG[18017]:
update_user_counter(804) - decrement outUse counter ...
Jun 12 17:44:32 DEBUG[18041]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18019]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18021]: Building dynamic conference '999'
Jun 12 17:44:32 DEBUG[18028]: update_user_counter(404) - decrement
outUse counter Jun 12 17:44:32 DEBUG[18042]: Placed channel
SIP/401-1bec in ZAP conf
1023
Jun 12 17:44:32 DEBUG[18043]: Placed channel SIP/601-d011
in ZAP conf
1023
Jun 12 17:44:32 DEBUG[18043]: Hangup: channel: -2 index =
0, normal =
41, callwait = -1, thirdcall = -1 Jun 12 17:44:32 DEBUG[18043]: Set
Re: [Asterisk-Users] home routers
Shaun, I believe that there are 2 models of the WRT54GP2 as there was/is with the PAP2's one that is set for VONAGE and one that is not, typically referred to as the WRT54GP2-NA John M On Monday June 19 2006 3:37 pm, Shaun wrote: I'm looking for somehting like the standard house hold linksys/dlink router. Basically it needs to have at least 1x100mbit port, wireless G capabilitys and at least 1 x anolog voip/sip connection. I've found linksys's WRT54GP2 which appears to do what i want. Anybody use this? Does it require vontage's service? I'm looking for any recommendations. 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] Re: DTMF Talk off
Matt, Thank you very much! I am currently running 1.2.7.1 but will be upgrading to 1.2.9.1 this week. I will try toneduration=200 first and let you/list know how well it worked. I read in zapata.conf.sample where it says: How long generated tones (DTMF and MF) will be played on the channel (in milliseconds) and did not realize that would have an effect on recognition. Thanks again, John M On Monday June 19 2006 2:58 pm, Matt King wrote: With recent versions of *, you can increase the detection time in zapata.conf with the toneduration variable. The default setting is: toneduration=100 We had the same problem and solved it by increasing this to 200. You can also increase the threshold volume for detection of DTMF by setting VPM_DEFAULT_DTMFTHRESHOLD in the relevant zaptel wctX.c and recompiling (though if you increase this too much you risk losing your ability to detect DTMF at all). Hope this helps, Matt. ___ --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] Help with Audicodes MP-104
Not quite sure. Audiocodes gives a dialtone when the number is called from PSTN. After few seconds I see the SIP invite to the Asterisk box. Asterisk responds with SIP 404 .Thanks,Lal On 6/12/06, Erick Perez [EMAIL PROTECTED] wrote: So is the problem with your audiocodes or with the asterisk system?if it is with the asterisk, what kind of calls are you trying route toyour box? SIP/IAX/other?On 6/12/06, Mahilal Silva [EMAIL PROTECTED] wrote: Hi All I have been able to get MP 104 FXO to make outbound calls with my asterisk box and polycom IP 500 phone. However I cannot get the incoming calls to hit the asterisk box. Any help will be appreciated. Thanks, Lal ___ --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 --Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de Panama Cel Panama. +(507) 6694-4780___--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
