Re: [Asterisk-Users] PCI Master Abort
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Beaumont Sent: Monday, July 07, 2003 4:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PCI Master Abort I am always getting multiple PCI Master Abort messages when I try to plug in a second TDM400P. I have asked this question before, but I nothing really solved my problem and I just put it on the back burner for a while. I am at a point where this is a crucial issue. I do have the same PCI Master abort message with a Wildcard S400P It seems this is NOT an IRQ problem : I did change the IRQ in the BIOS : - manual assignation for all the PCI boards - automatic assignation for the Wildcard then verificate the status of the PCI devices : - more /proc/pci - just to list the IRQ really assigned, the memory I/O addresses, ... All seems to be correct Then launch the modules : modprobe zaptel modprobe wcfxo modprobe wcfxs But here the error message still appears ! I'm interested in your solution if you solve the pb ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] msn
hi guys, have any of you guys managed to usemsn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the register message, at this point, i'm quite stuck. i also noticed that the nonce field isranddata?compared to iptel.org's ser and vovida's vocal. i notice a lot of difference on the sip messages composition. i'm running 0.4.0.
Re: [Asterisk-Users] Virtual fax on the Asterisk box
Hi Jim, Thank you for your detailed answer. I doubt anything is available (yet) that speaks iax/iax2, but sip or h.323 should be supported. Just make sure it can take a g.711 call and act like a fax machine; and take a bitmap and generate a g.711 call to send that document. If all you can find is a g.711 to t.38 solution, openh323 has support for t.38 to eg hylafax. A bit of a rube-goldberg, but it ought to work. I think to a SIP or IAX user agent ( I don't want to use H.323) with only G.711 support and able to emulate a virtual modem who then be used by a standard FAX application (Linux or Windows based) The biggest issue with these kinds of setups is latency. The fax protocols, as you can imagine, have latency requriements that are more stringent than voice calls. I need it only on the LAN, with a PSTN connection through a X100P card, so latency must not be an issue. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] msn
At 14:23 8-7-2003 +0800, you wrote: hi guys, have any of you guys managed to use msn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the register message, at this point, i'm quite stuck. i also noticed that the nonce field is randdata? compared to iptel.org's ser and vovida's vocal. i notice a lot of difference on the sip messages composition. i'm running 0.4.0. There was someting about using [EMAIL PROTECTED] as the sign-in name to indicate the proper realm, but you would have to experiment a little, since I fail to remember the exact details - sorry :P Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Switch issues with non-dedicated comms.. (My experience)
Hi, NOTE: This mail is sent for general information based on my experience to the Asterisk community not as an attack on Asterisk which I think is an amazing system.. Some of you may have seen my posting a few days ago where I was having problems with using the switch command in my dialplan.. It was suggested to put the switch command into its own [context] and then to include it into the contexts where it was required.. While using switch in the context that contains all my extensions or using it in a seperate context and then including it both worked fine on my setup this had not solved the issue I was having. After more testing I would simply advise that anyone who does not have a dedicated (and somewhat reliable) link between their two systems simply don't use the switch command and rather configure a wildcard extension mapping to the remote Asterisk system.. The problems I experienced is that when the link between the two servers is down the local server becomes highly unreliable.. Firstly when dialing an outside line it attempts to search the remote (at this time unavailable) dialplan before it has processed all the local options, This causes a massive delay between the time the call is dialed and the time the call actually starts the connection.. I have tried qualify=yes in the iax.conf but this did not alter the call processing routine in any way or reduce the time it takes from the time the number is dailed to the time it finds the extention mapping it needs to use.. The second issue is that when the line is down and an outbound call is placed (on my local system it is using an X100P) the call will terminate after approx 1-2 mins for no apparent reason.. I can only assume this has somthing to do with Asterisk trying to test connectivity with the remote system or somthing else to do with the switch command becasue when I comment out the switch command and use wildcard extension mappings the problem is gone.. so effectively what I have done is replaced this.. switch = IAX2/... with this.. exten = _2xxx,1,IAX2/... etc.. Now I am sure many of you are going to respond and tell me that my dialplan is wrong or that you are using switch and its perfect.. Thats all fine and I agree that switch is perfect when the link between the two systems is up.. All I am saying is that one day when you link is down for whatever reason and suddenly your local pbx starts to freak out and has long delay when placing a call or disconnects you constantly then rememebr my experience.. It will save you many hours of stress.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conferences with CAPI and H323
Hi; I try to work with conferences room by MeetMe and ztdummy modules. I notice that Conference with phones work well. But conference with H323 terminals work but there no sound. And conferences between H323 and capi only CAPI can talk and hear something. Does anyone has successfully make this ? Regards Rattana
FW: [Asterisk-Users] ATA 186 in Australia
The details for the Australian cisco ATA186 are below: -Original Message- From: Tony Du [mailto:[EMAIL PROTECTED] Sent: Tuesday, 8 July 2003 4:31 PM To: 'Adam Goryachev' Subject: RE: [Asterisk-Users] ATA 186 in Australia Hi Adam, I sold a Cisco ATA186 I1 2 port adaptor (Cisco code: SW-SMH-UL-ATA-2P)to you on 16/10/02) Cheers Tony Du -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Goryachev Sent: Tuesday, 8 July 2003 1:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ATA 186 in Australia Talk to Tony from Action Computers in Sydney (02) 9281 3988 and tell him I sent you, or else LAN Systems (Denis Valente) or any Tech Pacific reseller... I bought one from Tony, but I don't remember which one I ended up with. The easy way to tell is that one model is 'in stock' while the other was a 8 week lead time for me... Regards, Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Honson Sent: Tuesday, 8 July 2003 9:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ATA 186 in Australia Hi All, I'm looking at setting up a Asterisk system, and hope to use ATA 186's with it. Im in Australia, and am getting mixed answers to if its the I1 or I2 i need, does anyone have any experience with using ATA 186's in Australia Also, can anyone recommend a good place to obtain these locally? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Switch issues with non-dedicated comms.. (My experience)
The problems I experienced is that when the link between the two servers is down the local server becomes highly unreliable.. Firstly when dialing an outside line it attempts to search the remote (at this time unavailable) dialplan before it has processed all the local options, This causes a massive delay between the time the call is dialed and the time the call actually starts the connection.. I have tried qualify=yes in the iax.conf but this did not alter the call processing routine in any way or reduce the time it takes from the time the number is dailed to the time it finds the extention mapping it needs to use.. Perhaps something similar to the qualify for SIP clients could be implemented with IAX and IAX2 connections, ie, if ping time is greater than qualify time, then don't use it. It would also help if periodically asterisk would collect the extensions list for each switch statement included. This might cause minor problems if your extensions change frequantly (perhaps rather than just a periodic check include support to 'notify' remote servers when the local dialplan is changed like the BIND DNS support for zone files). I have only just started using IAX connections over a 64k ISDN link, and it is bearable as long as there is no other traffic using the link at the same time... I am yet to really play with it and make it work fantastically. Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfert call
Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana
Re: [Asterisk-Users] asterisk and uclinux
Hello,every one! I would like to know if asterisk could run under uclinux. look at the archives in jun-2003, we spoke about how to compile in under uclibc - there is a patch. but i personaly had problem to run it, because of i cannot run asterisk a as a daemon ( fork ).. i thing beacuse uClinux can only do vfork(). cheers Marian Regards. -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfert call
Sorry I did a mistake!! the correct way for the extension line is: exten =111,1,dial,Zap/1|20|T or exten =111,1,dial(Zap/1,20,T) I did a mix... cmayor --- Rattana BIV [EMAIL PROTECTED] escribió: Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chanh323 dialling
what is the format for an h323 entry in the dialplan? can I use chan_h323 without compiling anything else or should I compile oh323? basically what's the best way :) cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ECHO on sip- call
Hi all. Just got my 'Developer Kit Lite', installed it, and made the changes to load the modules in kernel and in the configuration files. Call thru-from fxo and the fxs sound great. Even fxs-iax-sip sound ok. When a answer a call coming into asterisk from the PSTN thru the fxo i have a loud echo. I have echocancel=yes. Is there another parameter I can change in the configuration files ? or do I have to change in... don't remember the name of the source program, the 'echo cancellation' to strong and recompile asterisk again? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re. rtp.c RTP codec 19
hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get Invalid CSeq Number back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chanh323 dialling
--- Dave Alan Caruana [EMAIL PROTECTED] escribió: what is the format for an h323 entry in the dialplan? [o]H323/exten@host for example: exten=21,1,dial(H323/[EMAIL PROTECTED]) when you dial this extension Asterisk call to the extension 11 on the host 192.192.192.192. can I use chan_h323 without compiling anything else Yes, you can, you don't need anything else or should I compile oh323? Only use ONE of them, h323 or oh323. If you want to use oh323 you must compile it. basically what's the best way :) Both goes ok, it's your decision. cheers Dave regards cmayor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323 v0.5.3
Hello all, I have updated the asterisk-oh323 package. The new version has several improvements (fixes in audio/RTP stream generation, music-on-hold working, flash hook detection, more config options). You can download it from: http://www.inaccessnetworks.com/projects/asterisk-oh323 Feedback is always welcome. Regards, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] three way calling and cisco ata 186
Pavel Zheltouhov wrote: Ok, if this is not working with sip or h.323, maybe it does with mgcp ? I tried to get ATA and Asterisk working with MGCP, but nothing worked! Any Howtos available about MGCP/ATA186/Asterisk? Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and VMWare
I use a similar asterisk setup when my T100P card is being used outside the home. I ran it on a 1ghz athlon that was shared as my workstation, but all linux. When my screen saver started, the audio quality dropped below usable. So obviously my system was idle if the screen saver started. SO as you can see you most likely will not get good results under vmware even if you are on a pretty hefty machine. On Tue, 2003-07-08 at 01:43, Dan wrote: Hi Gary, This possibility is excluded because the Home Automation framework runing on WinXP now needs to have direct access to a couple of proprietary hardware devices. I think to have a home Asterisk box on the same computer without any specific hardware, just to be able to use a couple of SIP phones (Cisco 7960 and ATA 186) without GSM support. The PSTN connection is made at the office, through another Asterisk box connected to the home one using IAX and GSM as codec. BR, Dan - Original Message - From: Gary Gapinski [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 07, 2003 10:55 PM Subject: Re: [Asterisk-Users] Asterisk and VMWare On Monday 07 July 2003 15:26, Dan wrote: The reason I ask this is because I have a Win2K PC running 24/7 which has enough power left, but if I cannot use any of the Digium hardware from inside VMWare then is useless. If you have not yet purchased the VMware license, run Linux, Asterisk, VMware _for Linux_, and W2K within VMware (with the caveat that special hardware will likely not be supported within VMs). Even if you have already purchased VMware for Microsoft Windows (the MS and Linux licenses are not interchangeable), contact VMware regarding a possible exchange. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answering on an zap device
How can I accept calls on a Wildcard E400P . Please include the zaptel.conf , zapata.conf and extension.conf to fully understand everything. Take a look firts to my configuration file. Where I did wrong? zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 # Span 1 bchan=1-15,17-31 dchan=16 # Span 2 bchan=32-46,48-62 dchan=47 # Span 3 bchan=63-77,79-93 dchan=78 # Span 4 bchan=94-108,110-124 dchan=109 alaw=1-124 loadzone = nl defaultzone=nl zapata.conf: [channels] context=inbound switchtype=euroisdn signalling=pri_cpe ;rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=no rxgain=0.0 txgain=0.0 immediate=yes ; Span 1 group=1 context=inbound signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 context=inbound signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 context=outbound signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 context=outbound signalling=pri_cpe channel = 94-108 channel = 110-124 and extensions.conf: [general] static=yes writeprotect=yes ... [inbound] exten = 1,1,Ringing exten = 1,2,Wait,2 exten = 1,3,Playback(beep) exten = 1,4,Playback(agent-alreadyon) exten = _XXX,1,Ringing exten = _XXX,2,Wait,2 exten = _XXX,3,Playback(beep) exten = #,1,Hangup ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent in new CVS
I installed the 7/7/03 CVS today, and my customer service reps
said that there were problems. So I went back to an earlier version.
They could log in, but when they received a call, they hear
the beep, but not the announcement or call.
I am using macros, if night is on, I dial the 800# that is passed.
Otherwise I play the Thank You (ty_pn), and I also pass the
announcement that the agent hears (pillnetwork).
Is this something in the latest CVS that may be a problem?
===
exten = 1086,1,Macro(enqueue,ty_pn,pillnetwork,18009159222)
[macro-enqueue]
exten = s,1,Wait,1
exten = s,2,DBGet($Night=GlobalSettings/Night)
exten = s,3,GotoIf($[${ARG3}]?4:5)
exten = s,4,Dial(Zap/g1/${ARG3}|120|t)
exten = s,5,VoiceMail,u300
exten = s,103,PlayBack,${ARG1}
exten = s,104,PlayBack,please_hold
exten = s,105,Queue(PillNetwork|t||${ARG2})
___
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Re: [Asterisk-Users] Answering on an zap device
First off, a E1 circuit is 32 channels, 1-32, 33-64, 65-96, 97-128. Also there is a D channel on 16 and 32 of each span. Then you need to add the D channel definition to your zapata.conf files. I believe all this was covered in the examples. On Tue, 2003-07-08 at 08:20, Cristi wrote: How can I accept calls on a Wildcard E400P . Please include the zaptel.conf , zapata.conf and extension.conf to fully understand everything. Take a look firts to my configuration file. Where I did wrong? zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 # Span 1 bchan=1-15,17-31 dchan=16 # Span 2 bchan=32-46,48-62 dchan=47 # Span 3 bchan=63-77,79-93 dchan=78 # Span 4 bchan=94-108,110-124 dchan=109 alaw=1-124 loadzone = nl defaultzone=nl zapata.conf: [channels] context=inbound switchtype=euroisdn signalling=pri_cpe ;rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=no rxgain=0.0 txgain=0.0 immediate=yes ; Span 1 group=1 context=inbound signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 context=inbound signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 context=outbound signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 context=outbound signalling=pri_cpe channel = 94-108 channel = 110-124 and extensions.conf: [general] static=yes writeprotect=yes ... [inbound] exten = 1,1,Ringing exten = 1,2,Wait,2 exten = 1,3,Playback(beep) exten = 1,4,Playback(agent-alreadyon) exten = _XXX,1,Ringing exten = _XXX,2,Wait,2 exten = _XXX,3,Playback(beep) exten = #,1,Hangup ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP.C codec error 19
hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get Invalid CSeq Number back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave (I mistakenly put an re in the title of this email and I think it's been ignored .. reposted) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 problem (small one)
can you please send your oh323.conf and your zapata.conf? cmayor --- Dave Alan Caruana [EMAIL PROTECTED] escribió: I have just compiled installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait(Zap/1-1, 1) in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial(Zap/1-1, OH323/[EMAIL PROTECTED]) in new stack 1:04.782 ThreadID=0x4958a540 H323 Attempt to use invalid URL [EMAIL PROTECTED]:1720 -- Couldn't call [EMAIL PROTECTED] -- Hungup 'H323:0' == Everyone is busy at this time -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' my extensions.conf is : [incoming] exten = s,1,Wait,1 exten = s,2,Dial(OH323/[EMAIL PROTECTED]) exten = s,3,Hangup I can't see anything obviously wrong, and dialling that h323 address from SJphone works fine (and used to work fine from asterisk too before). help please :) Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering on an zap device
Steven Critchfield wrote: First off, a E1 circuit is 32 channels, 1-32, 33-64, 65-96, 97-128. Also there is a D channel on 16 and 32 of each span. Then you need to add the D channel definition to your zapata.conf files. I believe all this was covered in the examples. On Tue, 2003-07-08 at 08:20, Cristi wrote: How can I accept calls on a Wildcard E400P . Please include the zaptel.conf , zapata.conf and extension.conf to fully understand everything. Take a look firts to my configuration file. Where I did wrong? zaptel.conf: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3 span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 # Span 1 bchan=1-15,17-31 dchan=16 # Span 2 bchan=32-46,48-62 dchan=47 # Span 3 bchan=63-77,79-93 dchan=78 # Span 4 bchan=94-108,110-124 dchan=109 alaw=1-124 loadzone = nl defaultzone=nl zapata.conf: [channels] context=inbound switchtype=euroisdn signalling=pri_cpe ;rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=no rxgain=0.0 txgain=0.0 immediate=yes ; Span 1 group=1 context=inbound signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 context=inbound signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 context=outbound signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 context=outbound signalling=pri_cpe channel = 94-108 channel = 110-124 and extensions.conf: [general] static=yes writeprotect=yes ... [inbound] exten = 1,1,Ringing exten = 1,2,Wait,2 exten = 1,3,Playback(beep) exten = 1,4,Playback(agent-alreadyon) exten = _XXX,1,Ringing exten = _XXX,2,Wait,2 exten = _XXX,3,Playback(beep) exten = #,1,Hangup ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users What is the need for the callerid in the following configuration: (default config page) ;callerid=Green Phone(256) 428-6121 ;channel = 1 ;callerid=Black Phone(256) 428-6122 ;channel = 2 ;callerid=CallerID Phone (256) 428-6123 ;callerid=CallerID Phone (630) 372-1564 ;callerid=CallerID Phone (256) 704-4666 ;channel = 3 ;callerid=Pac Tel Phone (256) 428-6124 ;channel = 4 ;callerid=Uniden Dead (256) 428-6125 ;channel = 5 ;callerid=Cortelco 2500 (256) 428-6126 ;channel = 6 ;callerid=Main TA 750 (256) 428-6127 ;channel = 44 It is the mapping between PTSN nr and channel numbers? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ENUM lookups
enum.diff Description: Binary data For those interested, this improves the ways lookups for NAPTR records are done. Maybe it can be patched on the main source tree by the authors. Other thing I bumped into was that compiling with gcc 2.95 cause asterisk to segfault when doing ENUM lookups. Compiling with gcc 3.3 solved this (on debian testing). Adrian Georgescu AG Projects [EMAIL PROTECTED] www.ag-projects.com IP phone: sip:[EMAIL PROTECTED] Managed DNS for IP telephony http://managed-dns.org/
Re: [Asterisk-Users] Transfert call
That got implemented recently ... Martin On Tue, 8 Jul 2003, carlos del mayor wrote: Hi Rattana, That kind of transfer is not yet implemented in *. The way it will be indicated is: exten =111,dial,Zap/1,20,T The T indicate that transfer is permitted for calling party, but as I've said, that's not implemented at the moment. Regards cmayor --- Rattana BIV [EMAIL PROTECTED] escribió: Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lot's of errors and warnings.
On Monday 07 July 2003 08:28 pm, Steven Critchfield wrote: Do you have the source that your kernel was compiled from? Do you at least have the appropriate headers for you kernel and the config file that was used? Haven't a clue. I was just following the install instructions as per :- http://www.asterisk.org/index.php?menu=download -- Neighbors!! We got neighbors! We ain't supposed to have any neighbors, and I just had to shoot one. -- Post Bros. Comics ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP.C codec error 19
Um no. Turn off Silence suppression (VAD) on your endpoint. Jeremy McNamara Lord Stroud wrote: Hi Dave, The RTP codec 19 error that you are getting indicates that your endpoint is most probably activating silence supression, and that you are using a codec such as g.729, at least, that is what I get on my platform here. You can go into the the rtc.c file, and simply comment out the message. Edit the rtp.c file at line 330, as the following: ast_log(LOG_NOTICE, Unknown RTP codec %d received\n, payloadtype); simply edit it to be: //ast_log(LOG_NOTICE, Unknown RTP codec %d received\n, payloadtype); and simply re-compile. Nir S On Tuesday 08 July 2003 04:42 pm, Dave Alan Caruana wrote: hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get Invalid CSeq Number back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave (I mistakenly put an re in the title of this email and I think it's been ignored .. reposted) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lot's of errors and warnings.
On Tue, 2003-07-08 at 09:34, marrandy wrote: On Monday 07 July 2003 08:28 pm, Steven Critchfield wrote: Do you have the source that your kernel was compiled from? Do you at least have the appropriate headers for you kernel and the config file that was used? Haven't a clue. I was just following the install instructions as per :- http://www.asterisk.org/index.php?menu=download Look at the last section. I'll reproduce it here so you know what I'm talking about. Note that your system MUST meet these requirements: You must have readline and openssl and their respective development packages. You must be running Linux 2.4.x You must have the Linux Kernel Sources package installed on your system. Look closely at the last line. If you followed instructions, then you would have had a clue. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lot's of errors and warnings.
On Tuesday 08 July 2003 10:34 am, marrandy wrote: On Monday 07 July 2003 08:28 pm, Steven Critchfield wrote: Do you have the source that your kernel was compiled from? Do you at least have the appropriate headers for you kernel and the config file that was used? Haven't a clue. I was just following the install instructions as per :- http://www.asterisk.org/index.php?menu=download Aahh...looks like the linux kernel source wasn't installed on this machine (Mandrake Linux v9.1). Done... Builds fine now. Thread end. -- People of privilege will always risk their complete destruction rather than surrender any material part of their advantage. -- John Kenneth Galbraith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] overlap dialing on a pri span
Well first of all if you set up DigitTimeout to 5 seconds so asterisk is going to wait up to 5 seconds to retrieve the digits specially when you have a match of _X. that is at least to digits but with the timeout of 5 you could imagine that asterisk will intercept all digits. How about having a pattern _X (without a dot). The amount of digits that asterisk is waiting for is set by you. _X is one digit, _X is 5 digits Martin On Tue, 8 Jul 2003 [EMAIL PROTECTED] wrote: Martin, I probably should have mentioned that: overlapdial=yes was set in zapata.conf (I take it this option is inherited through all the channels I configure in zapata.conf). I also did a fresh checkout today. My guess is that the main problem for now lies in the fact that asterisk won't execute a dial application once it received the first digit. Apparently, the extension _X. won't spawn dial before asterisk hits the timeout: exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,2 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,2 ; Set Response Timeout to 10 seconds exten = _X.,1,Dial,Zap/g8/BYEXTENSION I can see asterisk pick up: -- Executing Answer(Zap/159-1, ) in new stack the receive some digits DEBUG[22551]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: 7 on Zap/159-1 DEBUG[22551]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: 8 on Zap/159-1 ... and seconds (!) later asterisk dials out -- Executing Dial(Zap/159-1, Zap/g8/BYEXTENSION) in new stack -- Called g8/78997899 -- Channel 1, span 8 got hangup Do you know why? Is there a minimum number of digits asterisk need for an inital setup message? Thilo overlapdial=yes in zapata.conf for those channels that you want the overlapdialing be activated. By default only incoming overlap dialing is enabled. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering on an zap device
Steven Critchfield wrote: Please trim unnecessary lines from the post. On Tue, 2003-07-08 at 09:23, Cristi wrote: What is the need for the callerid in the following configuration: (default config page) ;callerid=Green Phone(256) 428-6121 ;channel = 1 ;callerid=Black Phone(256) 428-6122 ;channel = 2 ;callerid=CallerID Phone (256) 428-6123 ;callerid=CallerID Phone (630) 372-1564 ;callerid=CallerID Phone (256) 704-4666 ;channel = 3 ;callerid=Pac Tel Phone (256) 428-6124 ;channel = 4 ;callerid=Uniden Dead (256) 428-6125 ;channel = 5 ;callerid=Cortelco 2500 (256) 428-6126 ;channel = 6 ;callerid=Main TA 750 (256) 428-6127 ;channel = 44 It is the mapping between PTSN nr and channel numbers? This is to make internal phones show callerID. Notice the Green phone, Black Phone, These are to specify what callerid to use when the channel is used to place a call. In my office and at home I have set each channels caller ID to specify the user at the end of the phone. This way you know who it is inside your office calling you when you see the callerid. Also, when used appropriately, you can specify either your main trunk number or specific DID numbers for the callerID display on a PRI line. This way you could set up a 9+ dialing for DID display, and 8+ for main line display. This is helpful when you are doing generic calling or specific calling. First let me thank you for all the answers that I got from the asterisk people! Another question : Let say that now I have a E1 ISDN into a 4E1 card and I don't have any errors :. What configuration I have to do get a call ? Tring a nr associated with the channel I got only a busy line!From where the asterisk know what nr to answer ? It is specified into the extension config file for the context associated with the channel group? == D-Channel on span 1 up == D-Channel on span 3 up -- B-channel 1 successfully restarted on span 3 -- B-channel 2 successfully restarted on span 3 -- B-channel 3 successfully restarted on span 3 -- B-channel 4 successfully restarted on span 3 -- B-channel 5 successfully restarted on span 3 -- B-channel 6 successfully restarted on span 3 -- B-channel 7 successfully restarted on span 3 -- B-channel 8 successfully restarted on span 3 -- B-channel 9 successfully restarted on span 3 -- B-channel 10 successfully restarted on span 3 -- B-channel 11 successfully restarted on span 3 -- B-channel 12 successfully restarted on span 3 -- B-channel 13 successfully restarted on span 3 -- B-channel 14 successfully restarted on span 3 -- B-channel 15 successfully restarted on span 3 -- B-channel 17 successfully restarted on span 3 -- B-channel 18 successfully restarted on span 3 -- B-channel 19 successfully restarted on span 3 -- B-channel 20 successfully restarted on span 3 -- B-channel 21 successfully restarted on span 3 -- B-channel 22 successfully restarted on span 3 -- B-channel 23 successfully restarted on span 3 -- B-channel 24 successfully restarted on span 3 -- B-channel 25 successfully restarted on span 3 -- B-channel 26 successfully restarted on span 3 -- B-channel 27 successfully restarted on span 3 -- B-channel 28 successfully restarted on span 3 -- B-channel 29 successfully restarted on span 3 -- B-channel 30 successfully restarted on span 3 -- B-channel 31 successfully restarted on span 3 -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 -- B-channel 3 successfully restarted on span 1 -- B-channel 4 successfully restarted on span 1 -- B-channel 5 successfully restarted on span 1 -- B-channel 6 successfully restarted on span 1 -- B-channel 7 successfully restarted on span 1 -- B-channel 8 successfully restarted on span 1 -- B-channel 9 successfully restarted on span 1 -- B-channel 10 successfully restarted on span 1 -- B-channel 11 successfully restarted on span 1 -- B-channel 12 successfully restarted on span 1 -- B-channel 13 successfully restarted on span 1 -- B-channel 14 successfully restarted on span 1 -- B-channel 15 successfully restarted on span 1 -- B-channel 17 successfully restarted on span 1 -- B-channel 18 successfully restarted on span 1 -- B-channel 19 successfully restarted on span 1 -- B-channel 20 successfully restarted on span 1 -- B-channel 21 successfully restarted on span 1 -- B-channel 22 successfully restarted on span 1 -- B-channel 23 successfully restarted on span 1 -- B-channel 24 successfully restarted on span 1 -- B-channel 25 successfully restarted on span 1 -- B-channel 26 successfully restarted on span 1 -- B-channel 27 successfully restarted on span 1 -- B-channel 28 successfully restarted on span 1 -- B-channel 29 successfully restarted on span 1 -- B-channel 30 successfully restarted on span 1 -- B-channel 31 successfully restarted on span 1
[Asterisk-Users] Using multiple iconnecthere accounts
Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=_91NXXNXX,1,StripMSD,1 exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED];iconnect is the first account exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED];iconnect2 is the second account But that doesn't work. Has anybody tried this before? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 problem (small one)
Hi, Dave Alan Caruana wrote: I have just compiled installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait(Zap/1-1, 1) in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial(Zap/1-1, OH323/[EMAIL PROTECTED]) in new stack 1:04.782 ThreadID=0x4958a540 H323Attempt to use invalid URL [EMAIL PROTECTED]:1720 -- Couldn't call [EMAIL PROTECTED] -- Hungup 'H323:0' == Everyone is busy at this time -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' There are some changes in the OpenH323 library regarding the calling scheme used. The new style is h323:[EMAIL PROTECTED] I thought that it would be better not to make changes on the called address before making the call (adding this h323: prefix), in order to be more flexible. Of course this breaks older extension files (only in the case of a direct call, something like [EMAIL PROTECTED]). my extensions.conf is : [incoming] exten = s,1,Wait,1 exten = s,2,Dial(OH323/[EMAIL PROTECTED]) So, just change it into OH323/h323:[EMAIL PROTECTED] and you should be just fine. exten = s,3,Hangup I can't see anything obviously wrong, and dialling that h323 address from SJphone works fine (and used to work fine from asterisk too before). help please :) Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RTP.C codec error 19
Hey JJ, You are correct, as always. However, please remember that some carriers will simply say: Sorry, if you don't support VAD, we are not going to change our setup for a single client. Regards, Nir S On Tuesday 08 July 2003 05:44 pm, Jeremy McNamara wrote: Um no. Turn off Silence suppression (VAD) on your endpoint. Jeremy McNamara Lord Stroud wrote: Hi Dave, The RTP codec 19 error that you are getting indicates that your endpoint is most probably activating silence supression, and that you are using a codec such as g.729, at least, that is what I get on my platform here. You can go into the the rtc.c file, and simply comment out the message. Edit the rtp.c file at line 330, as the following: ast_log(LOG_NOTICE, Unknown RTP codec %d received\n, payloadtype); simply edit it to be: //ast_log(LOG_NOTICE, Unknown RTP codec %d received\n, payloadtype); and simply re-compile. Nir S On Tuesday 08 July 2003 04:42 pm, Dave Alan Caruana wrote: hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get Invalid CSeq Number back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave (I mistakenly put an re in the title of this email and I think it's been ignored .. reposted) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lot's of errors and warnings.
On Tuesday 08 July 2003 10:47 am, Steven Critchfield wrote: Yes Steven, I realized after I posted the last mail that I had only checked the first two parts of the MUST requirement and had assumed the kernel src was installed. mea culpa ZNot yet sure why it wasn't installed on this machine. Will have to look into that. -- When it comes to broken marriages most husbands will split the blame -- half his wife's fault, and half her mother's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfert call
Sorry! Didn't know it got implemented!!Last notice I had is that it would be implemented soon, but didn't think it was SO soon...Great then!! cmayor --- Martin Pycko [EMAIL PROTECTED] escribió: That got implemented recently ... Martin On Tue, 8 Jul 2003, carlos del mayor wrote: Hi Rattana, That kind of transfer is not yet implemented in *. The way it will be indicated is: exten =111,dial,Zap/1,20,T The T indicate that transfer is permitted for calling party, but as I've said, that's not implemented at the moment. Regards cmayor --- Rattana BIV [EMAIL PROTECTED] escribió: Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using multiple iconnecthere accounts
Did asterisk register with both accounts ? sip show registry Can you post what happens on the console along with 'sip debug' ? Martin On Tue, 8 Jul 2003, Derek Beaumont wrote: Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=_91NXXNXX,1,StripMSD,1 exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED] ;iconnect is the first account exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED] ;iconnect2 is the second account But that doesn't work. Has anybody tried this before? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answering on an zap device
On Tue, 2003-07-08 at 10:06, Cristi wrote: Steven Critchfield wrote: Please trim unnecessary lines from the post. On Tue, 2003-07-08 at 09:23, Cristi wrote: What is the need for the callerid in the following configuration: (default config page) ;callerid=Green Phone(256) 428-6121 ;channel = 1 ;callerid=Black Phone(256) 428-6122 ;channel = 2 ;callerid=CallerID Phone (256) 428-6123 ;callerid=CallerID Phone (630) 372-1564 ;callerid=CallerID Phone (256) 704-4666 ;channel = 3 ;callerid=Pac Tel Phone (256) 428-6124 ;channel = 4 ;callerid=Uniden Dead (256) 428-6125 ;channel = 5 ;callerid=Cortelco 2500 (256) 428-6126 ;channel = 6 ;callerid=Main TA 750 (256) 428-6127 ;channel = 44 It is the mapping between PTSN nr and channel numbers? This is to make internal phones show callerID. Notice the Green phone, Black Phone, These are to specify what callerid to use when the channel is used to place a call. In my office and at home I have set each channels caller ID to specify the user at the end of the phone. This way you know who it is inside your office calling you when you see the callerid. Also, when used appropriately, you can specify either your main trunk number or specific DID numbers for the callerID display on a PRI line. This way you could set up a 9+ dialing for DID display, and 8+ for main line display. This is helpful when you are doing generic calling or specific calling. First let me thank you for all the answers that I got from the asterisk people! Another question : Let say that now I have a E1 ISDN into a 4E1 card and I don't have any errors :. What configuration I have to do get a call ? Tring a nr associated with the channel I got only a busy line!From where the asterisk know what nr to answer ? It is specified into the extension config file for the context associated with the channel group? Turn on debug for your pri spans so you can see how many digits are being sent to you in the called number part of the q.931 packets. From there you will need to specify in extensions.conf a extension for each of the phone numbers associated with the PRIs to the precision of what you observed before. Good chance is that it is the full number. As for the context, you should have specified that in your zapata.conf file as to where these calls would be dropped into. I have mine defined as PRI and then reroute appropriately from there. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] line battery check
I know in the x100p hardware the driver can sense whether there is battery voltage on the line or not. Is this possible in the zap drivers using t1/e1 interfaces ? I believe the signalling bits are only ring and hook, not loop presence, so not sure if it is possible. I had a situation last night where a main cable was cut in the area and lines were down overnight. My burglar alarm knew right away there was no line battery and started beeping. Data T1 down was a no-brainer for the monitoring software to detect. Now I am looking for a way to detect loop loss on the voice lines - any ideas how to accomplish with asterisk ? I have a zhone bank - anyone know offhand if I can program loop loss on a port to put it into an alarm state ? At least then I would have a contact closure I could monitor easily. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lot's of errors and warnings.
On Tue, 2003-07-08 at 10:15, marrandy wrote: On Tuesday 08 July 2003 10:47 am, Steven Critchfield wrote: Yes Steven, I realized after I posted the last mail that I had only checked the first two parts of the MUST requirement and had assumed the kernel src was installed. mea culpa ZNot yet sure why it wasn't installed on this machine. Will have to look into that. Because Mandrake is not a server distribution. Then again, I'm not sure of any of the package oriented system placing the kernel package on a machine by default. I guess most distributions assume if you are going to go through the effort of recompiling the kernel, then you can go get your kernel source. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using multiple iconnecthere accounts
Busy is n+1 if n+101 doesn't exist. Martin On 8 Jul 2003, Steven Critchfield wrote: On Tue, 2003-07-08 at 10:10, Derek Beaumont wrote: Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=_91NXXNXX,1,StripMSD,1 exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED];iconnect is the first account exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED];iconnect2 is the second account But that doesn't work. Has anybody tried this before? Isn't busy n+101 priority, or is it n+100? Basically you dial out similar to how you set up the busy portion of your voicemail. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] three way calling and cisco ata 186
Thomas Dingermann wrote: Ok, if this is not working with sip or h.323, maybe it does with mgcp ? I tried to get ATA and Asterisk working with MGCP, but nothing worked! Any Howtos available about MGCP/ATA186/Asterisk? I just try two ATA with asterisk with that configuration files : ; ; MGCP Configuration for Asterisk ; [general] port = 2727 bindaddr = 0.0.0.0 allow=ulaw inbanddtmf=yes transfer = yes threewaycalling=yes [10.0.1.19] transfer = yes threewaycalling=yes host = 10.0.1.19 context = default line = aaln/1 transfer = 1 line = aaln/2 transfer = 1 line = * [10.0.1.20] transfer = yes threewaycalling=yes host = 10.0.1.20 context = default line = aaln/1 transfer = 1 line = aaln/2 transfer = 1 line = * and extensions.conf --- exten = 31,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) exten = 32,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) exten = 33,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) exten = 34,1,Dial(MGCP/aaln/[EMAIL PROTECTED],20,tr) - Ordinary tasks works good. Call transfer with '#' key work too. But three way calling not work with stange error : -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down dial to 33 -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '3' -- Executing Dial(MGCP/aaln/[EMAIL PROTECTED], MGCP/aaln/[EMAIL PROTECTED]|20|tr) in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED]) -- MGCP cw: 0, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Called aaln/[EMAIL PROTECTED] -- MGCP/aaln/[EMAIL PROTECTED] is ringing answer -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP/aaln/[EMAIL PROTECTED] answered MGCP/aaln/[EMAIL PROTECTED] -- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED] -- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and MGCP/aaln/[EMAIL PROTECTED] Talking now Attempt call person 3 : hookflash : -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED] -- MGCP Muting 1 on aaln/[EMAIL PROTECTED] -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down now trying dial to other phone ( 600 - echo test ) -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '6' atfter this, person1 hear 'fastbusy', short beeps ! And other output of asterisk: -- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and MGCP/aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '0' -- Executing Playback(MGCP/aaln/[EMAIL PROTECTED], demo-echotest) in new stack -- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED] -- Playing 'demo-echotest' -- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and MGCP/aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hf' -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED] -- We didn't make one of the calls FLIPFLOP 0 and 1 on aaln/[EMAIL PROTECTED] -- MGCP Muting 0 on aaln/[EMAIL PROTECTED] -- Attempting native bridge of MGCP/aaln/[EMAIL PROTECTED] and MGCP/aaln/[EMAIL PROTECTED] -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' NOTICE[20501]: File chan_mgcp.c, Line 762 (mgcp_fixup): mgcp_fixup(MGCP/aaln/[EMAIL PROTECTED], MGCP/aaln/[EMAIL PROTECTED]MASQ) WARNING[20501]: File chan_mgcp.c, Line 764 (mgcp_fixup): old channel wasn't 0x81065a8 but was (nil) WARNING[20501]: File channel.c, Line 1847 (ast_do_masquerade): Fixup failed on channel MGCP/aaln/[EMAIL PROTECTED]MASQ, strange things may happen. NOTICE[20501]: File chan_mgcp.c, Line 762 (mgcp_fixup): mgcp_fixup(MGCP/aaln/[EMAIL PROTECTED]ZOMBIE, MGCP/aaln/[EMAIL PROTECTED]) -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' == Spawn extension (default, 600, 1) exited non-zero on 'MGCP/aaln/[EMAIL PROTECTED]' -- Any ideas ? -- Pavel Zheltouhov, Comlink ISP, Voronezh, Russia phone/fax +7(0732) 727172, http://www.comlink.ru ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] line battery check
On Tue, 2003-07-08 at 10:29, Jon Pounder wrote: I know in the x100p hardware the driver can sense whether there is battery voltage on the line or not. Is this possible in the zap drivers using t1/e1 interfaces ? I believe the signalling bits are only ring and hook, not loop presence, so not sure if it is possible. I had a situation last night where a main cable was cut in the area and lines were down overnight. My burglar alarm knew right away there was no line battery and started beeping. Data T1 down was a no-brainer for the monitoring software to detect. Now I am looking for a way to detect loop loss on the voice lines - any ideas how to accomplish with asterisk ? I have a zhone bank - anyone know offhand if I can program loop loss on a port to put it into an alarm state ? At least then I would have a contact closure I could monitor easily. I don't think the zhones support external notification of alarm other than the LEDs. The CACs do for sure, and therefore it is a good bet the Adtran stuff does as well. I know on my CAC hardware there is an external plug to tie into to put any number of monitors on to detect when it has had a problem. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] line battery check
Hi- Its not clear from your msg whether you want to detect the open-circuit condition of the T1/E1 directly or on one of the analog circuits connected on the other side of the channel bank. If directly on the E1/T1, I imagine that asterisk can detect the condition of the alarm signals for the circuit - these are separate from the signalling supervision bits like on/off hook. I'm new to asterisk, so don't know how to read those alarm indicators in the software, but the ones you are probably interested in are: Red Alarm - means loss of signal (I think a cut line would cause this), Yellow alarm - means far end detected an error, and BPV - bi-polar violation, means degraded signal (usually). Hope this helps. regards Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Tuesday, July 08, 2003 4:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] line battery check I know in the x100p hardware the driver can sense whether there is battery voltage on the line or not. Is this possible in the zap drivers using t1/e1 interfaces ? I believe the signalling bits are only ring and hook, not loop presence, so not sure if it is possible. I had a situation last night where a main cable was cut in the area and lines were down overnight. My burglar alarm knew right away there was no line battery and started beeping. Data T1 down was a no-brainer for the monitoring software to detect. Now I am looking for a way to detect loop loss on the voice lines - any ideas how to accomplish with asterisk ? I have a zhone bank - anyone know offhand if I can program loop loss on a port to put it into an alarm state ? At least then I would have a contact closure I could monitor easily. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using multiple iconnecthere accounts
--- Steven Critchfield [EMAIL PROTECTED] escribió: On Tue, 2003-07-08 at 10:10, Derek Beaumont wrote: Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=_91NXXNXX,1,StripMSD,1 exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED] ;iconnect is the first account exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED] ;iconnect2 is the second account But that doesn't work. Has anybody tried this before? Isn't busy n+101 priority, or is it n+100? When you use the dial application, if the interface you call to is busy, asterisk goes to priority n+101 if it's exist. If there is no response or priority n+101 doesn't exist, then goes to priority n+1 as usually. Basically you dial out similar to how you set up the busy portion of your voicemail. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Messenger - Nueva versión GRATIS Super Webcam, voz, caritas animadas, y más... http://messenger.yahoo.es ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using multiple iconnecthere accounts
Asterisk has registered with both accounts: sip show registry Host Username Refresh State 213.137.73.178:5060 120 Registered 213.137.73.178:5060 120 Registered I can make one call just fine, but when I try to make the second call, I get an invalid extension error. When using the following configuration: exten=_91NXXNXX,1,StripMSD,1 exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED] exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED] I get the following output Executing Dial(Zap/4-1, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnect-cd45 is making progress passing it to Zap/4-1 show channels Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter Format 213.137.73.178 xx 6631b1e766b 00103/0 0ms ms 4 1 active SIP channel(s) This appears when I make the first call. I notice that I have a 0ms Jitter buffer. I am now curious as to how I create a jitter buffer in sip.conf? I have the following in the [general] section of sip.conf jitterbuffer=yes dropcount=3 maxjitterbuffer=2500 maxexccessbuffer=100 Below is the output when I tried to call a second long distance number -- Executing Dial(Zap/4-2, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 480 Temporarily not available back from 213.137.73.178 -- SIP/iconnect-fde9 is circuit-busy == Everyone is busy at this time -- Executing Dial(Zap/4-2, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] sip show channels Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter Format 213.137.37.178 xx 7047ee1a76b 00102/0 0ms ms 2 213.137.73.176 xx 7b782a7b3dd 00103/0 0ms ms 4 2 active SIP channel(s) *CLI WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) == No one is available to answer at this time -- Sent into invalid extension 'xxx' in context 'outgoing' on Zap/4-2 -- Executing Playback(Zap/4-2, TelError) in new stack -- Playing 'TelError' WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Any help is appreciated. Thank you for your time. OLD MESSAGE=== Did asterisk register with both accounts ? sip show registry Can you post what happens on the console along with 'sip debug' ? Martin On Tue, 8 Jul 2003, Derek Beaumont wrote: Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=_91NXXNXX,1,StripMSD,1 exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED] ;iconnect is the first account exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED] ;iconnect2 is the second account But that doesn't work. Has anybody tried this before? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lot's of errors and warnings.
On Tuesday 08 July 2003 11:15 am, marrandy wrote: Spoke too soon. Mamdrake v9.1 - hopefully, this will help other people who scan the arcvives first. cd ../asterisk # make clean ; make install - asterisk.c: In function `cli_complete': asterisk.c:833: warning: assignment of read-only location bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: bison: Command not found make: *** [ast_expr.c] Error 127 checked and installed bison. now makes fine. Added make samples. But make progdocs fails with :- [EMAIL PROTECTED] asterisk]# make progdocs doxygen asterisk-ng-doxygen make: doxygen: Command not found make: *** [progdocs] Error 127 [EMAIL PROTECTED] asterisk]# -- The price one pays for pursuing any profession, or calling, is an intimate knowledge of its ugly side. -- James Baldwin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Accounting
Why doesn't the CDR show outgoing numbers? I need a record of outbound digits dialed to reconcile my phone bills. __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debug PRI!
This indicate that the connection with the local provider PTSN it is ok? : -- Attempting call on Zap/10 for [EMAIL PROTECTED]:1 (Retry 2) -- Making new call for cr 32781 Protocol Discriminator: Q.931 (8) len=28 Call Ref: len= 2 (reference 13/0xD) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] Display (len= 1) [ 1 ] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] Called Number (len= 5) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '10' ] Sending Complete (len= 0) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32781/0x800D) (Terminator) Message type: RELEASE COMPLETE (90) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Normal, unspecified (31), class = Normal Event (1) ] -- Processing IE 8 (Cause) -- Channel 10, span 1 got hangup -- Hungup 'Zap/10-1' NOTICE[28690]: File pbx_spool.c, Line 195 (attempt_thread): Call failed to go through, reason 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lot's of errors and warnings.
On Wednesday, July 9, 2003, at 12:15 AM, marrandy wrote: [about the kernel sources on Mandrake 9.1] ZNot yet sure why it wasn't installed on this machine. Will have to look into that. This would seem to be a Mandrake thing. I had the same problem. The Mandrake installer would not install the kernel sources when doing a CD based installation. Although the kernel sources are on CD3, the package manager interface doesn't seem to find/show them. I did an FTP based install instead in order to get a complete system with kernel sources. rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] line battery check
At 05:18 PM 7/8/2003 +0100, you wrote: Hi- Its not clear from your msg whether you want to detect the open-circuit condition of the T1/E1 directly or on one of the analog circuits connected on the other side of the channel bank. the t1 circuit is fine and the channel bank did not alarm during the problem (it does have a form c alarm contact) What I am asking is if anyone knows offhand if the alarm state can be triggered on the fxo/fxs ports by reprogramming, or if it only applies to the t1. OR is there any way with rbs to detect a line that lost loop power ? OR is there a way to have asterisk check every few minutes when a line is not in use that it can raise a dialtone ? if not then have some other process get triggered. If directly on the E1/T1, I imagine that asterisk can detect the condition of the alarm signals for the circuit - these are separate from the signalling supervision bits like on/off hook. I'm new to asterisk, so don't know how to read those alarm indicators in the software, but the ones you are probably interested in are: Red Alarm - means loss of signal (I think a cut line would cause this), Yellow alarm - means far end detected an error, and BPV - bi-polar violation, means degraded signal (usually). Hope this helps. regards Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Tuesday, July 08, 2003 4:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] line battery check I know in the x100p hardware the driver can sense whether there is battery voltage on the line or not. Is this possible in the zap drivers using t1/e1 interfaces ? I believe the signalling bits are only ring and hook, not loop presence, so not sure if it is possible. I had a situation last night where a main cable was cut in the area and lines were down overnight. My burglar alarm knew right away there was no line battery and started beeping. Data T1 down was a no-brainer for the monitoring software to detect. Now I am looking for a way to detect loop loss on the voice lines - any ideas how to accomplish with asterisk ? I have a zhone bank - anyone know offhand if I can program loop loss on a port to put it into an alarm state ? At least then I would have a contact closure I could monitor easily. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using multiple iconnecthere accounts
Derek, tray this - it's working 100% with iconnect:
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Best regards
Lubo
Derek Beaumont wrote:
Asterisk has registered with both accounts:
sip show registry
Host Username Refresh State
213.137.73.178:5060 120 Registered
213.137.73.178:5060 120 Registered
I can make one call just fine, but when I try to make the second call,
I get an invalid extension error. When using the following
configuration:
exten=_91NXXNXX,1,StripMSD,1
exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]
I get the following output
Executing Dial(Zap/4-1, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/iconnect-cd45 is making progress passing it to Zap/4-1
show channels
Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter
Format
213.137.73.178 xx 6631b1e766b 00103/0 0ms ms
4
1 active SIP channel(s)
This appears when I make the first call. I notice that I have a 0ms
Jitter buffer. I am now curious as to how I create a jitter buffer
in sip.conf? I have the following in the [general] section of sip.conf
jitterbuffer=yes
dropcount=3
maxjitterbuffer=2500
maxexccessbuffer=100
Below is the output when I tried to call a second long distance number
-- Executing Dial(Zap/4-2, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily not available back from
213.137.73.178
-- SIP/iconnect-fde9 is circuit-busy
== Everyone is busy at this time
-- Executing Dial(Zap/4-2, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
sip show channels
Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter
Format
213.137.37.178 xx 7047ee1a76b 00102/0 0ms ms
2
213.137.73.176 xx 7b782a7b3dd 00103/0 0ms ms
4
2 active SIP channel(s)
*CLI WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Request)
== No one is available to answer at this time
-- Sent into invalid extension 'xxx' in context 'outgoing'
on Zap/4-2
-- Executing Playback(Zap/4-2, TelError) in new stack
-- Playing 'TelError'
WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for
seqno 102 (Request)
Any help is appreciated. Thank you for your time.
OLD MESSAGE===
Did asterisk register with both accounts ?
sip show registry
Can you post what happens on the console along with 'sip debug' ?
Martin
On Tue, 8 Jul 2003, Derek Beaumont wrote:
Has anybody out there tried to use two different iconnecthere accounts
with Asterisk?
What I want to do is use a second account if the first is busy.
I have tried the following:
exten=_91NXXNXX,1,StripMSD,1
exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED] ;iconnect is the
first account
exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED] ;iconnect2 is
the second account
But that doesn't work. Has anybody tried this before?
___
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RE: [Asterisk-Users] line battery check
There's got to be a way... I think zttool shows a red alarm on an X100P when there's no phone line plugged into it (and I would guess when there's no voltage on the line.) My guess is that it gets the info from /proc/zap-something-or-other, but I'm just guessing. Jared On Tue, 2003-07-08 at 11:12, Jon Pounder wrote: At 05:18 PM 7/8/2003 +0100, you wrote: Hi- Its not clear from your msg whether you want to detect the open-circuit condition of the T1/E1 directly or on one of the analog circuits connected on the other side of the channel bank. the t1 circuit is fine and the channel bank did not alarm during the problem (it does have a form c alarm contact) What I am asking is if anyone knows offhand if the alarm state can be triggered on the fxo/fxs ports by reprogramming, or if it only applies to the t1. OR is there any way with rbs to detect a line that lost loop power ? OR is there a way to have asterisk check every few minutes when a line is not in use that it can raise a dialtone ? if not then have some other process get triggered. If directly on the E1/T1, I imagine that asterisk can detect the condition of the alarm signals for the circuit - these are separate from the signalling supervision bits like on/off hook. I'm new to asterisk, so don't know how to read those alarm indicators in the software, but the ones you are probably interested in are: Red Alarm - means loss of signal (I think a cut line would cause this), Yellow alarm - means far end detected an error, and BPV - bi-polar violation, means degraded signal (usually). Hope this helps. regards Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Tuesday, July 08, 2003 4:29 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] line battery check I know in the x100p hardware the driver can sense whether there is battery voltage on the line or not. Is this possible in the zap drivers using t1/e1 interfaces ? I believe the signalling bits are only ring and hook, not loop presence, so not sure if it is possible. I had a situation last night where a main cable was cut in the area and lines were down overnight. My burglar alarm knew right away there was no line battery and started beeping. Data T1 down was a no-brainer for the monitoring software to detect. Now I am looking for a way to detect loop loss on the voice lines - any ideas how to accomplish with asterisk ? I have a zhone bank - anyone know offhand if I can program loop loss on a port to put it into an alarm state ? At least then I would have a contact closure I could monitor easily. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lot's of errors and warnings.
Hadn't finished and sent the last mail by mistake. Anyway... Mandrake v9.1 #make progdocs But make progdocs fails with :- -- [EMAIL PROTECTED] asterisk]# make progdocs doxygen asterisk-ng-doxygen make: doxygen: Command not found make: *** [progdocs] Error 127 [EMAIL PROTECTED] asterisk]# - checked and added doxygen. Now builds successfully. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debug PRI!
On Tue, 2003-07-08 at 12:06, Cristi wrote: This indicate that the connection with the local provider PTSN it is ok? : Called Number (len= 5) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '10' ] This is the interesting line. Does the number your calling to get into this system end in 10? if so it would seem you are only getting the last 2 digits of the number and need to set up an extension for 10 in the context you are dropping the call into. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Accounting
On Tue, 2003-07-08 at 12:07, Erik Kendall wrote: Why doesn't the CDR show outgoing numbers? I need a record of outbound digits dialed to reconcile my phone bills. What technology are you using? I have no problem with recording the outbound numbers called on my zap channels. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Accounting
I'm using Zap. TDM10P and X100P Thanks for any help. --- Erik Kendall [EMAIL PROTECTED] wrote: Why doesn't the CDR show outgoing numbers? I need a record of outbound digits dialed to reconcile my phone bills. __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Accounting
are you using some exten to get an external dialtone? like 0 - give a dialtone then compose the number In this way numbers could not be logged, since are dialled 'natively' on the telco dialtone by the user, not by asterisk. Matteo. Il mar, 2003-07-08 alle 19:07, Erik Kendall ha scritto: Why doesn't the CDR show outgoing numbers? I need a record of outbound digits dialed to reconcile my phone bills. __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budgetone and Voicemail
I have a problem with using voicemail on the Budgetone phones. When entering the mailbox and password, sometimes some keys will register multiple times (as shown on console when it says no such user in config file) and sometimes some keys won't even register at all. It seems totally random. Has anyone seen this problem? Any recommendations would be greatly appreciated. Thanks. Brian Borders [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Budgetone and Voicemail
Yes I have seen it. I had to change the digit time in the voicemail app and recompile. There is a new voicemail2 app. I have not used it, but maybe it fixes this problem. If you test it out, let me know how it works for you. Brian Borders wrote: I have a problem with using voicemail on the Budgetone phones. When entering the mailbox and password, sometimes some keys will register multiple times (as shown on console when it says no such user in config file) and sometimes some keys won't even register at all. It seems totally random. Has anyone seen this problem? Any recommendations would be greatly appreciated. Thanks. Brian Borders [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using multiple iconnecthere accounts
How about that:
exten = _91NXXNXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]SIP/${EXTEN:[EMAIL
PROTECTED]
Martin
On Tue, 8 Jul 2003, Derek Beaumont wrote:
Asterisk has registered with both accounts:
sip show registry
Host Username Refresh State
213.137.73.178:5060 120 Registered
213.137.73.178:5060 120 Registered
I can make one call just fine, but when I try to make the second call,
I get an invalid extension error. When using the following
configuration:
exten=_91NXXNXX,1,StripMSD,1
exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]
I get the following output
Executing Dial(Zap/4-1, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/iconnect-cd45 is making progress passing it to Zap/4-1
show channels
Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter
Format
213.137.73.178 xx 6631b1e766b 00103/0 0ms ms
4
1 active SIP channel(s)
This appears when I make the first call. I notice that I have a 0ms
Jitter buffer. I am now curious as to how I create a jitter buffer
in sip.conf? I have the following in the [general] section of sip.conf
jitterbuffer=yes
dropcount=3
maxjitterbuffer=2500
maxexccessbuffer=100
Below is the output when I tried to call a second long distance number
-- Executing Dial(Zap/4-2, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily not available back from
213.137.73.178
-- SIP/iconnect-fde9 is circuit-busy
== Everyone is busy at this time
-- Executing Dial(Zap/4-2, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
sip show channels
Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter
Format
213.137.37.178 xx 7047ee1a76b 00102/0 0ms ms
2
213.137.73.176 xx 7b782a7b3dd 00103/0 0ms ms
4
2 active SIP channel(s)
*CLI WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Request)
== No one is available to answer at this time
-- Sent into invalid extension 'xxx' in context 'outgoing'
on Zap/4-2
-- Executing Playback(Zap/4-2, TelError) in new stack
-- Playing 'TelError'
WARNING[98311]: File chan_sip.c, Line 410 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for
seqno 102 (Request)
Any help is appreciated. Thank you for your time.
OLD MESSAGE===
Did asterisk register with both accounts ?
sip show registry
Can you post what happens on the console along with 'sip debug' ?
Martin
On Tue, 8 Jul 2003, Derek Beaumont wrote:
Has anybody out there tried to use two different iconnecthere accounts
with Asterisk?
What I want to do is use a second account if the first is busy.
I have tried the following:
exten=_91NXXNXX,1,StripMSD,1
exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED];iconnect is the
first account
exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED];iconnect2 is
the second account
But that doesn't work. Has anybody tried this before?
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Re: [Asterisk-Users] Budgetone and Voicemail
I have had double digits being passed every now and then once I am into voicemail.. I haven't had a problem with the initial login stage.. I also haven't had time to look into it yet.. You could try changing the DTMF mode and see if it helps.. Later.. I have a problem with using voicemail on the Budgetone phones. When entering the mailbox and password, sometimes some keys will register multiple times (as shown on console when it says no such user in config file) and sometimes some keys won't even register at all. It seems totally random. Has anyone seen this problem? Any recommendations would be greatly appreciated. Thanks. Brian Borders [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Accounting
Yes, I'm using a 9. I tried the following found in sample extension.conf, but it didn't work because I have 10 digit local dialing. #exten = _9NXX,1,StripMSD,1 #exten = _NXX,2,Dial,Zap/1/BYEXTENSION So, I started using the following with success: #ignorepat = 9 #exten = 9,1,Dial,Zap/1/ What should I use for 10-digit local and 11-digit long distance dialing? Thanks, Erik --- Brancaleoni Matteo [EMAIL PROTECTED] wrote: are you using some exten to get an external dialtone? like 0 - give a dialtone then compose the number In this way numbers could not be logged, since are dialled 'natively' on the telco dialtone by the user, not by asterisk. Matteo. Il mar, 2003-07-08 alle 19:07, Erik Kendall ha scritto: Why doesn't the CDR show outgoing numbers? I need a record of outbound digits dialed to reconcile my phone bills. __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using multiple iconnecthere accounts
On Wednesday, July 9, 2003, at 02:46 AM, Derek Beaumont wrote:
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
What does EXTEN:1 do? Why is StripMSD not used?
EXTEN:1 expands into the extension dialed without the first digit,
that's why you don't need StripMSD.
rgds
bk
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Re: [Asterisk-Users] Budgetone and Voicemail
using speakerphone with dtmf=inband? that's caused by a too loud dtmf tones when on speakerphone. upgrade to latest firmware (.72) and use dtmf=info or rfc2833, should to the trick. Matteo. Il mar, 2003-07-08 alle 20:21, Brian Borders ha scritto: I have a problem with using voicemail on the Budgetone phones. When entering the mailbox and password, sometimes some keys will register multiple times (as shown on console when it says no such user in config file) and sometimes some keys won't even register at all. It seems totally random. Has anyone seen this problem? Any recommendations would be greatly appreciated. Thanks. Brian Borders [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BudgeTone-100 Early Dial
Hi Stephen Thanks for the reply I am using inband DTMF and firmware version 1.0.3.72 Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen R. Besch Sent: 07 July 2003 23:50 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] BudgeTone-100 Early Dial Paul, First, make sure that you use inband DTMF. As far as I know, out of band still does not work. Second, make sure that the firmware is up to date. The silent DTMF problem was fixed a few releases ago (at rev xx.xx.xx.60 I believe). -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Accounting
That's the reason why no number is logged, since
you simply connect the user to the telco, and let him
dial. To have the number logged, asterisk must
dial, so you can use this exten matching:
exten = _9.,2,Dial(Zap/1/${EXTEN:1})
the '.' after the nine just match anything you dial.
after the timeout (3 or five secs, don't remind),
asterisk dials the entire number to Zap/1 ,
stripping away the lead digit 9 , since EXTEN:1
means 'what have you dialled, beside the first digit'
Matteo.
Il mar, 2003-07-08 alle 20:55, Erik Kendall ha scritto:
Yes, I'm using a 9. I tried the following found in
sample extension.conf, but it didn't work because I
have 10 digit local dialing.
#exten = _9NXX,1,StripMSD,1
#exten = _NXX,2,Dial,Zap/1/BYEXTENSION
So, I started using the following with success:
#ignorepat = 9
#exten = 9,1,Dial,Zap/1/
What should I use for 10-digit local and 11-digit long
distance dialing?
Thanks,
Erik
--- Brancaleoni Matteo [EMAIL PROTECTED] wrote:
are you using some exten to get an external
dialtone?
like 0 - give a dialtone
then compose the number
In this way numbers could not be logged, since
are dialled 'natively' on the telco dialtone by the
user,
not by asterisk.
Matteo.
Il mar, 2003-07-08 alle 19:07, Erik Kendall ha
scritto:
Why doesn't the CDR show outgoing numbers? I need
a
record of outbound digits dialed to reconcile my
phone bills.
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[Asterisk-Users] Using multiple iconnecthere accounts
First off, sorry for using a mail client without the in-reply-to
function.
Second: I still can't make two calls using iconnecthere at the same
time.
Here is what I have tried:
Attempt 1:
exten=_91NXXNXX,1,Dial,StripMSD
exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]
Attempt 2:
exten=_91NXXNXX,1,Dial,StripMSD
exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
exten=_1NXXNXX,103,Dial,SIP/[EMAIL PROTECTED]
Attempt 3:
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Attempt 4:
exten =
_91NXXNXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]SIP/${EXTEN:[EMAIL PROTECTED]
So far nothing has worked.
Another question I have is about jitter buffer. Is there a way to
create a
Jitter buffer in sip.conf?
When I type sip show channels I get the following output:
sip show channels
Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter
Format
213.137.73.176 xx 5752cb7a55f 00103/0 0ms ms
4
There is a section for Jitter, so I would imagine that there is some way
to do it.
Thank you for your time.
Also, if anybody could suggest a good mail client for windows that is
able to
use the in-reply-to function, it would be helpful.
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Re: [Asterisk-Users] Using multiple iconnecthere accounts
On your place I would check separately if you can use both accounts. I
think that one of your accounts in disabled ...
Martin
On Tue, 8 Jul 2003, Derek Beaumont wrote:
First off, sorry for using a mail client without the in-reply-to
function.
Second: I still can't make two calls using iconnecthere at the same
time.
Here is what I have tried:
Attempt 1:
exten=_91NXXNXX,1,Dial,StripMSD
exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
exten=_1NXXNXX,3,Dial,SIP/[EMAIL PROTECTED]
Attempt 2:
exten=_91NXXNXX,1,Dial,StripMSD
exten=_1NXXNXX,2,Dial,SIP/[EMAIL PROTECTED]
exten=_1NXXNXX,103,Dial,SIP/[EMAIL PROTECTED]
Attempt 3:
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Attempt 4:
exten =
_91NXXNXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]SIP/${EXTEN:[EMAIL PROTECTED]
So far nothing has worked.
Another question I have is about jitter buffer. Is there a way to
create a
Jitter buffer in sip.conf?
When I type sip show channels I get the following output:
sip show channels
Peer UsernameCall ID Seq (Tx/Rx) Lag Jitter
Format
213.137.73.176 xx 5752cb7a55f 00103/0 0ms ms
4
There is a section for Jitter, so I would imagine that there is some way
to do it.
Thank you for your time.
Also, if anybody could suggest a good mail client for windows that is
able to
use the in-reply-to function, it would be helpful.
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Re: [Asterisk-Users] Call Accounting
Matteo,
Thank you. I'll give it a try.
Erik
--- Brancaleoni Matteo [EMAIL PROTECTED] wrote:
That's the reason why no number is logged, since
you simply connect the user to the telco, and let
him
dial. To have the number logged, asterisk must
dial, so you can use this exten matching:
exten = _9.,2,Dial(Zap/1/${EXTEN:1})
the '.' after the nine just match anything you dial.
after the timeout (3 or five secs, don't remind),
asterisk dials the entire number to Zap/1 ,
stripping away the lead digit 9 , since EXTEN:1
means 'what have you dialled, beside the first
digit'
Matteo.
Il mar, 2003-07-08 alle 20:55, Erik Kendall ha
scritto:
Yes, I'm using a 9. I tried the following found
in
sample extension.conf, but it didn't work because
I
have 10 digit local dialing.
#exten = _9NXX,1,StripMSD,1
#exten = _NXX,2,Dial,Zap/1/BYEXTENSION
So, I started using the following with success:
#ignorepat = 9
#exten = 9,1,Dial,Zap/1/
What should I use for 10-digit local and 11-digit
long
distance dialing?
Thanks,
Erik
--- Brancaleoni Matteo [EMAIL PROTECTED]
wrote:
are you using some exten to get an external
dialtone?
like 0 - give a dialtone
then compose the number
In this way numbers could not be logged, since
are dialled 'natively' on the telco dialtone by
the
user,
not by asterisk.
Matteo.
Il mar, 2003-07-08 alle 19:07, Erik Kendall ha
scritto:
Why doesn't the CDR show outgoing numbers? I
need
a
record of outbound digits dialed to reconcile
my
phone bills.
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Re: [Asterisk-Users] Budgetone and Voicemail
Hi, The same problem with a Cisco 7960 phone too. I don't think is phone related. BR, Dan - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 08, 2003 9:42 PM Subject: Re: [Asterisk-Users] Budgetone and Voicemail I have had double digits being passed every now and then once I am into voicemail.. I haven't had a problem with the initial login stage.. I also haven't had time to look into it yet.. You could try changing the DTMF mode and see if it helps.. Later.. I have a problem with using voicemail on the Budgetone phones. When entering the mailbox and password, sometimes some keys will register multiple times (as shown on console when it says no such user in config file) and sometimes some keys won't even register at all. It seems totally random. Has anyone seen this problem? Any recommendations would be greatly appreciated. Thanks. Brian Borders [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 prob :)
i'm getting Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait(Zap/1-1, 1) in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial(Zap/1-1, OH323/h323:[EMAIL PROTECTED]) in new stack 5:59.330 H323 Cleaner H323Connection ip$localhost/18729 terminated. ERROR[1230546240]: File chan_oh323.c, Line 704 (oh323_call): H323:0: Could not call h323:[EMAIL PROTECTED] -- Couldn't call h323:[EMAIL PROTECTED] -- Hungup 'H323:0' == Everyone is busy at this time -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' any idea what that can mean ? I have my system currently working through SIP, however every now and then it shows this message -- Got SIP response 481 Invalid CSeq Number back from 216.52.153.207 == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1' and drops the line which is the reason I am trying to use H323 instead, maybe I can get around that problem. Can anyone tell me what it means? thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP disconnecting : response 481
-- Got SIP response 481 Invalid CSeq Number back from 216.52.153.207 == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1' I am getting this error on an outgoing call to a SIP host. The call just disconnects .. is there any way around it ? thanks Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Problem (previous post) .. information might be relevant
regarding my previous post about SIP outgoing calls dropping with an error 481 .. this is my output from a SIP debug. the call dropped occurs at the end. Asterisk is mine, Cisco-SIPGateway is the other end (remote) and not in my control. help :) please!! Dave Signal=0 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Date: Tue, 08 Jul 2003 22:22:57 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 117 INFO Contact: sip:[EMAIL PROTECTED]:5060 10 headers, 0 lines set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 118 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=1 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Date: Tue, 08 Jul 2003 22:22:58 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 118 INFO Contact: sip:[EMAIL PROTECTED]:5060 10 headers, 0 lines set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 119 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=2 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Date: Tue, 08 Jul 2003 22:22:58 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 119 INFO Contact: sip:[EMAIL PROTECTED]:5060 10 headers, 0 lines set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=250 (no NAT) to 216.52.153.207:5060 set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 121 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=6 Duration=250 (no NAT) to 216.52.153.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Date: Tue, 08 Jul 2003 22:22:59 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 121 INFO Contact: sip:[EMAIL PROTECTED]:5060 10 headers, 0 lines Retransmitting #1 (no NAT): INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=250 to 216.52.153.207:5060 set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: 21382890 sip:[EMAIL PROTECTED];tag=as6556b0d9 To: sip:[EMAIL PROTECTED];tag=26845C24-FDA Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 122 INFO User-Agent: Asterisk PBX Content-Type: application/dtmf-relay Content-Length: 24 Signal=4 Duration=250 (no
Re: [Asterisk-Users] Please help -- Syntax for dialing VoIP provider
Hi BK, Using your configuration info, I now have Nikotel working again. Other than the fromuser=, it appears that one also now needs the auth=md5 whereas before it was not necessary. To disable incoming calling, just delete the register - line for Nikotel. That way, no one can find you. You do not need to the register - line for outgoing calls. On Monday, July 7, 2003, at 11:16 AM, BK [address only for mailing lists] wrote: Hi Paul, thanks for your insights On Monday, July 7, 2003, at 03:59 PM, Paul Cheng wrote: To dial a PSTN number through Nikotel used to work from Asterisk, but they had a very serious security issue (you could make calls anytime anywhere and their billing wouldn't charge it) and after I informed them of this, they changed their authentication mechanism and since then I have not gotten it to work (they didn't even thank me!). This is what we have discovered last night. However, We have got it working now. I will document this in detail and make it available, but briefly here a quick summary ... First I had various glitches in my dial string. With the help of John Todd and some others on the IRC #asterisk channel I was able to fix those glitches. Thanks everybody who assisted. Then I tried a number of things I had already experimented with before. When I turned on SIP debug and watched the datagrams, I could see Nikotel's response account name does not match address of record. Together with the from part, this led me to fiddle with fromuser again and when I set it to the actual login name, it worked. Their tech people said it should work with a slight change: yes, we changed it yesterday. Now the user part of the From: address has to be the same as the username in the Proxy-Authentication line. I don't know if the Asterisk can do that. The ATA186 does it b[y] default. This CAN be done if you edit chan_sip.c, It would seem you can do it a lot simpler: in sip.conf --- -- register = myusername:[EMAIL PROTECTED] [nikotel] username=myusername fromuser=myusername ... --- -- but when I did this, it billed me a few times for unconnected calls Thanks for sharing this with us. I will watch this for a while and see if this happens here too. and I gave up trying to debug and switched to iConnect. iConnect is worse quality, but it is very easy to connect to. I had much better quality with calls via Nikotel than iConnect, but their support is non-existent/bad at best. I sent them 3-4 e-mails about their security issue before they even responded. Yes, support is not exactly their strength, is it?! FYI. Registering with Nikotel was futile anyways, because I never figured out how anyone could call into me. I don't want anybody to call in via Nikotel. Since they do not provide a telephone number for incoming calls, the only calls you could possibly get are from their public chat room. In the very best case you get a friendly test call from somebody who has just signed up and wants to try out the service, in the worst case you get prank calls in the middle of the night or indecent proposals and all the rest of it. I will have to find a way to disable incoming calls from Nikotel entirely. iConnect provides a PSTN-SIP dial in as an option, but I haven't tried it. Yes, I have seen that. And at $8.95/mth it would seem reasonably priced, too. Outbound calls do not require registering. I can provide examples of iConnect connection scripts if you contact me offline. Thanks, I will do that. again many thanks to everybody who has helped solving this riddle rgds bk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Paul Cheng Mátyás király ut 10 H-1121 Budapest HUNGARY [EMAIL PROTECTED] mobile: +36 30 381-9311 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTEL toll-free
Hi, Has anyone been able to place a call via IAXTEL toll-free termination lately? I had it working at one time, but now it doesn't seem to work anymore. www.iaxtel.com also appears dead. Is this the server problem again or is it my config? Haven't been able to find any references in the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codec problems with asterisk
We appear to be having a problem with our asterisk setup. We have a cisco AS5300 with pri lines coming in and passing the calls onto asterisk then too the sip phones. the phone call from the sip phones (7960's) appears to be ok nice and clear including the user who has called in. but if your the user who has called in its all crackley sounds really bad when they speak. i believe this problem is a codec problem as far as i can see we use ulaw across the board, the 5300 currently supports 12 different codecs however asterisk only like too work with ulaw or alaw it tends to not except the call if the other codecs are used. clear-channel Clear Channel 64000 bps g711alaw G.711 A Law 64000 bps g711ulaw G.711 u Law 64000 bps g723ar53 G.723.1 ANNEX-A 5300 bps g723ar63 G.723.1 ANNEX-A 6300 bps g723r53G.723.1 5300 bps g723r63G.723.1 6300 bps g726r16G.726 16000 bps g726r24G.726 24000 bps g726r32G.726 32000 bps g728 G.728 16000 bps g729br8G.729 ANNEX-B 8000 bps g729r8 G.729 8000 bps ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P noise?
On Saturday 05 July 2003 10:18 pm, Kevin Herzig wrote: Hi all. I bought Digium's dev kit and a used IBM PL300 PC to try it out in. The X100P works fine, but with the TDM400P I get what I can best describe as 'interrupt noise'... noise whenever I type a key on the keyboard, or when something accesses the disk drive, uses cpu, etc. In my other PC it works fine, no noise and sounds great. But I cannot dedicate the other PC to it. Anyone have any ideas? I'd be grateful. So far I've had good success getting * to dial out via IConnectHere. Neat stuff! Kevin When I used * on an IBM PIII 600 (6565-85U) it also made a nice noise. I switched to a dell poweredge and it's gone. It seems to be related to the power supply. I'd try adding sime decent size capacitors (5000 u+) to 5 and 12V, if I had the time to mess with it. Check to see if it is certified for home use which is harder than office. -- Steve __ This sig is pending approval ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID number assignment to SIP phones
The concept is fuzzy... If one were to terminate a T1 PRI(s) into a T100P or T400P. How are DID numbers assigned to IP phones on the inside through a SIP gateway. What am I missing here?... -- Mark Street, D.C. Red Hat Certified Engineer Cert# 807302251406074 -- Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0 GPG key http://www.streetchiro.com/pubkey.asc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Accounting
You actually need to do
exten = _9NXXNXX,1,dial(Zap/1/${ETEN:1}) ; local
exten = _91NXXNXX,1,dial(Zap/1/${EXTEN:1}) ; LD
If you read the documentation and followed the examples you would
understand that the underscore starts a pattern, N is for digits 2-9, X
is for 0-9.
On Tue, 2003-07-08 at 14:55, Erik Kendall wrote:
Matteo,
Thank you. I'll give it a try.
Erik
--- Brancaleoni Matteo [EMAIL PROTECTED] wrote:
That's the reason why no number is logged, since
you simply connect the user to the telco, and let
him
dial. To have the number logged, asterisk must
dial, so you can use this exten matching:
exten = _9.,2,Dial(Zap/1/${EXTEN:1})
the '.' after the nine just match anything you dial.
after the timeout (3 or five secs, don't remind),
asterisk dials the entire number to Zap/1 ,
stripping away the lead digit 9 , since EXTEN:1
means 'what have you dialled, beside the first
digit'
Matteo.
Il mar, 2003-07-08 alle 20:55, Erik Kendall ha
scritto:
Yes, I'm using a 9. I tried the following found
in
sample extension.conf, but it didn't work because
I
have 10 digit local dialing.
#exten = _9NXX,1,StripMSD,1
#exten = _NXX,2,Dial,Zap/1/BYEXTENSION
So, I started using the following with success:
#ignorepat = 9
#exten = 9,1,Dial,Zap/1/
What should I use for 10-digit local and 11-digit
long
distance dialing?
Thanks,
Erik
--- Brancaleoni Matteo [EMAIL PROTECTED]
wrote:
are you using some exten to get an external
dialtone?
like 0 - give a dialtone
then compose the number
In this way numbers could not be logged, since
are dialled 'natively' on the telco dialtone by
the
user,
not by asterisk.
Matteo.
Il mar, 2003-07-08 alle 19:07, Erik Kendall ha
scritto:
Why doesn't the CDR show outgoing numbers? I
need
a
record of outbound digits dialed to reconcile
my
phone bills.
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[Asterisk-Users] RE: IAXTEL toll-free From: Asterisk-Users digest, Vol 1 #791 - 10 msgs
I asked on the IRC channel last night and was told the IAXTEL had been down for a few months now. It had a very poor uptime.. Maybe someone can tell us why the uptime was so poor. Alex Message: 9 Date: Wed, 9 Jul 2003 01:05:00 +0200 From: Paul Cheng [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAXTEL toll-free Reply-To: [EMAIL PROTECTED] Hi, Has anyone been able to place a call via IAXTEL toll-free termination lately? I had it working at one time, but now it doesn't seem to work anymore. www.iaxtel.com also appears dead. Is this the server problem again or is it my config? Haven't been able to find any references in the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need a recommendation on a good motherboard/processor combination
I seem to remember you sent this the other day? If so I`m guessing you didn`t get much replies, so... I need a recommendation on a good motherboard/processor combination. I would like a motherboard that has lots of PCI slots and works well with Asterisk without problems getting drivers working, etc. Onboard LAN would be nice to keep from using a slot. Plan to use RedHat 8 for the OS. I, and a reseller/consultant friend of mine, used to use ASUS a lot due to reliability. (note: I still have a working Asus P100 board running as a firewall here - probably run for a few years more fault free - they really used to be fantastic) In the last few years I personally have had way too much hassle with faulty boards (DOAs, firmware problems that never got fixed and intermittant faults specifically for me), and so did this friend of mine with his custumers, so we both will never touch them with a barge pole. Maybe they do score well on TH for speed, but I`m in business - I need reliability as a priority. I personally have and still do use Gigabyte and have never had a problem - though the way they have implemented dual bios defeats itself. (meaning a virus could concievably blow both bios chips away because they are both writable, and nvram is shared between the two) The last ASUS board I ever bought was extremely unreliable (strange intermittant boot problems among other things) so I replaced it with a Gigiabyte board with the same (VIA 333) chipset and it was solid as a rock (still using it now in fact) Gigabyte have a lot of boards with 5+ PCI slots, too. My aforementioned reseller friend is a fan of Intel boards due to reliability, just good design, and a _very_ good returns policy (very important when you are reselling to businesses) Well that`s how I see it - I`m sure others will dispute my claims, but they are purely experience-based. Hope this helps. Cheers, Mathew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P noise?
Thanks for the response, Steven. I've tried various configurations of with / without x and/or framebuffer. No difference. I'm running on Debian Woody with a kernel I compiled from the kernel source. Maybe I'll try to download the stock kernel source and compile that next. I've tried unmasking interrupts from the hard drive, with and without apm.o installed, etc, etc... It sounds to me like the same problem referenced here: http://www.mail-archive.com/[EMAIL PROTECTED]/msg02434.htm l The other end of the conversation hears the same noise I do. The conversation is intelligible, but barely. Too bad, I guess I'm going to have to get another machine to do this with. I can get another but it's getting to be expensive. :) Kevin On Sat, 2003-07-05 at 21:18, Kevin Herzig wrote: Hi all. I bought Digium's dev kit and a used IBM PL300 PC to try it out in. The X100P works fine, but with the TDM400P I get what I can best describe as 'interrupt noise'... noise whenever I type a key on the keyboard, or when something accesses the disk drive, uses cpu, etc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Attention TDM400P users (was Re: [Asterisk-Users] PCI Master Abort)
I believe the PCI master abort issue may also have something to do with power. As some of you know, we are currently working to resolve power issues on the TDM40B which have caused some people grief with poor audio or the PCI Master Abort syndrome. We (Wil and I) have hand modified a couple of boards to try to cleanup the PC power supply and are very pleased with the results (until just recently, we didn't have a machine with the noise problem in our hands). I believe our modifications represent a general panacea for a range of power supply related issues including: * PCI Master abort * Interrupt noise * Static * Buzzing Martin is making a list of people who have trouble with bad audio and/or PCI master abort. In the next couple of days, we would like to send some boards to people that have these problems *in the U.S.* and who are capable of installing and testing these cards in a *timely fashion*. If you are interested, please contact Martin (256-428-6161 or martinp) and be sure he has you on the list and knows what your symptom is and what your hardware configuration is. Once we have confirmed these fixes work properly, we will be spinning a revised version of the TDM400P board. Once the TDM400P replacement boards are ready, anyone who wants to upgrade will be able to exchange their old board for a new one at no cost (you will only be responsible for shipping us back your old board). If you are having no trouble with the board, there will be no need to send it back, although we will allow the boards to be exchanged for the forseeable future (so long as we produce anything like the TDM400P) so there will be no hurry in swapping out your board. Mark On Tue, 8 Jul 2003 [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Beaumont Sent: Monday, July 07, 2003 4:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PCI Master Abort I am always getting multiple PCI Master Abort messages when I try to plug in a second TDM400P. I have asked this question before, but I nothing really solved my problem and I just put it on the back burner for a while. I am at a point where this is a crucial issue. I do have the same PCI Master abort message with a Wildcard S400P It seems this is NOT an IRQ problem : I did change the IRQ in the BIOS : - manual assignation for all the PCI boards - automatic assignation for the Wildcard then verificate the status of the PCI devices : - more /proc/pci - just to list the IRQ really assigned, the memory I/O addresses, ... All seems to be correct Then launch the modules : modprobe zaptel modprobe wcfxo modprobe wcfxs But here the error message still appears ! I'm interested in your solution if you solve the pb ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voip
Hello. Well I now have asterisk installed. I've printed out the asterisk web site. I've printed the draft Asterisk handbook V2 I've printed the Introduction to the asterisk open source pbx Because I'm experimenting, I would like to do things in a certain order :- 1) VOIP inside the private LAN from one computer to another. e.g. 192.168.1.1 to 192.168.1.2 etc. 2) VOIP to someone outside in the U.S. 3) VOIP to someone overseas e.g. U.K. 4) Get a hardware card for the incoming line 5) Some extensions, perhap's Four (4). I don't see much in relation to point 1. What software (linux) can be used to connect VOIP to the astericks server ?? Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip
On Tue, 8 Jul 2003, marrandy wrote: Well I now have asterisk installed. I've printed out the asterisk web site. I've printed the draft Asterisk handbook V2 I've printed the Introduction to the asterisk open source pbx oh, no, the trees, the humanity... try and not print, its easier to cut paste from a console than print the buggers out, besides the pace of * development at times has been so fast, those printed pages are probably obsolete by now :) 1) VOIP inside the private LAN from one computer to another. e.g. 192.168.1.1 to 192.168.1.2 etc. great... IAX2 will do that for ya 2) VOIP to someone outside in the U.S. ask nufone, or setup your own gateway 3) VOIP to someone overseas e.g. U.K. ask nufone or taan or someone like that 4) Get a hardware card for the incoming line get an X100P from digium 5) Some extensions, perhap's Four (4). get a TDM400P from digium (or use soft phones) I don't see much in relation to point 1. why not? keep looking, it'll become clearer What software (linux) can be used to connect VOIP to the astericks server ?? sip|h323|iax|mgcp?|skinny? a whole buncha codecs - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P noise?
Then why not just get a new powersupply? If it is a standard ATX powersupply then it should be able to be replaced for $25-40. On Tue, 2003-07-08 at 19:41, Kevin Herzig wrote: Thanks for the response, Steven. I've tried various configurations of with / without x and/or framebuffer. No difference. I'm running on Debian Woody with a kernel I compiled from the kernel source. Maybe I'll try to download the stock kernel source and compile that next. I've tried unmasking interrupts from the hard drive, with and without apm.o installed, etc, etc... It sounds to me like the same problem referenced here: http://www.mail-archive.com/[EMAIL PROTECTED]/msg02434.htm l The other end of the conversation hears the same noise I do. The conversation is intelligible, but barely. Too bad, I guess I'm going to have to get another machine to do this with. I can get another but it's getting to be expensive. :) Kevin On Sat, 2003-07-05 at 21:18, Kevin Herzig wrote: Hi all. I bought Digium's dev kit and a used IBM PL300 PC to try it out in. The X100P works fine, but with the TDM400P I get what I can best describe as 'interrupt noise'... noise whenever I type a key on the keyboard, or when something accesses the disk drive, uses cpu, etc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
