[Asterisk-Users] Send parameters from asterisk to ADSI phone
Hi all ! I'm begginning with ADSI technology and I have some problem: I have a Aastra 390 and it seems to work well, I send my adsi script to the phone, i can call, display some messages, program new keys ... But, I don't find how to send an external parameters to the phone. Typically, I would like to display a value of an asterisk variable. I tried to find how the parameters $Call1p and $Call1s work, but I did not understant anything. Maybe if you have an great idea Sorry for my English! Thanks Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mini atx and asterisk (EPIA and the like)
The dual riser and fritz cards are unrelated. My production system is a C3 Samuel 5000, my development system is an M-1GHz Nemiah with the dual riser card. The dual riser problem is down to the dual riser and am dealing with Tranquil PC on it - although never tried it in the Samuel so thanks for the info. The Fritz problem is actually me not knowing enough about Linux to get the blasted thing working! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: 01 March 2005 22:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mini atx and asterisk (EPIA and the like) See my comments inline Razza wrote: I run mandrake 9.2, one FXO (x100p clone), 5 sip phones, MusicOnHold, voicemail, etc. off my EPIA Classic/5000 with 512MB memory (I know 512 is totally OTT but had a spare SD stick lying around after upgrading my main PC) and it works fine. I would like to also run a Fritz ISDN card but am unable to get this working in linux, also struggling with the dual active riser card (from Tranquil PC). Top PCI slot fails to do anything and bottom PCI slot is great - so avoid multiple cards until tranquil give some useful advice! The motherboard must support 2 slot PCI for it to work. IIRC, the Ezra-500/800 can't support 2 slots. The later M1 Nehemiah boards can support 2 slots. Fritz ISDN works beautifully with M1. My setup was Trustix 2.l + Fritz CAPI drivers + chan_capi. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR setup problems
Hi guys still have the problem to setup the IVR correctly.
I am forwarding call from ser :
if (method == "INVITE") { if (uri =~ "sip:[EMAIL PROTECTED]"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } }
inside sip.conf
-
port=5061bindaddr=0.0.0.0srvlookup=yes
[ser]type=peerhost=xxx.xxx.xxx.xxxcontext=ser1
inside extensions.conf
-
[ser1]Exten = 40,1,AnswerExten = 40,2,SetMusicOnHold(default)Exten = 40,3,DigitTimeout,5Exten = 40,4,ResponseTimeout,10Exten = 40,5,Background(greeting)
Exten = 1,1,Playback(secr) ; if you press 91192 playback message 93secr94Exten = 1,2,Dial(SIP/Phone1/20)
Exten = 2,1,Playback(studentservice)Exten = 2,2,Dial(SIP/Phone1/20)
Exten = 3,1,Playback(it)Exten = 3,2,Dial(SIP/Phone1/20)
Exten = 4,1,Playback(operator)Exten = 4,2,Dial(SIP/Phone1/20)
Inside asterisk debug i see what the forwarding of the call working :
log of ASTERISK DEBUG
Sip read: INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0Via: SIP/2.0/UDP ipoftphone:5060;branch=z9hG4bK06ffef7dFrom: "Alexg" sip:[EMAIL PROTECTED];tag=00036b09607e0047524bda98-4b96b81eTo: sip:[EMAIL PROTECTED]Call-ID: [EMAIL PROTECTED]CSeq: 101 INVITEUser-Agent: CSCO/6Contact: sip:[EMAIL PROTECTED]:5060Expires: 180Content-Type: application/sdpContent-Length: 249Accept: application/sdp
v=0o=Cisco-SIPUA 28416 11732 IN IP4 ipoftphones=SIP Callc=IN IP4 ipoftphonet=0 0m=audio 26298 RTP/AVP 0 8 18 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15
13 headers, 11 linesUsing latest request as basis requestSending to xxx.xxx.xxx.xxx : 5060 (non-NAT)Found peer 'ser'Found RTP audio format 0Found RTP audio format 8Found RTP audio format 18Found RTP audio format 101Peer audio RTP is at port ipoftphone:26298Found description format PCMUFound description format PCMAFound description format G729Found description format telephone-eventCapabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)Looking for 1phoneiamcalling in ser1Reliably Transmitting (no NAT):SIP/2.0 404 Not FoundVia: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0Via: SIP/2.0/UDP ipoftphone:5060;branch=z9hG4bK06ffef7dFrom: "Alexg" sip:[EMAIL PROTECTED];tag=00036b09607e0047524bda98-4b96b81eTo:
sip:[EMAIL PROTECTED];tag=as125ae8d3Call-ID: [EMAIL PROTECTED]CSeq: 101 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFERContact: sip:[EMAIL PROTECTED]:5061Content-Length: 0
to xxx.xxx.xxx.xxx:5060
Sip read: ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKb148.00624e85.0From: "Alexg" sip:[EMAIL PROTECTED];tag=00036b09607e0047524bda98-4b96b81eCall-ID: [EMAIL PROTECTED]To: sip:[EMAIL PROTECTED];tag=as125ae8d3CSeq: 101 ACKUser-Agent: Sip EXpress router(0.8.14 (i386/linux))Content-Length: 0
8 headers, 0 linesDestroying call '[EMAIL PROTECTED]'
var/log/asterisk/messages
---
Unable to open /dev/dsp: No such device
I am calling to number 122 and ser forwarding it to the asterisk (port 5061) (see configuration of sip.conf) to the ser1 context.
in extensions.conf i have ser1 context and extensions for ivr under ser1 context.
After the call i am hearing the busy line and that's it. i tried to play with extensions.conf with no success.
I need a help to setup the IVR system.
Thanks.
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[Asterisk-Users] Incorrect CDRs
Hello list, We are having some serious problems with CDR and billing. CDR shows that some of the (unanswered) calls were lasted for 2-3 days. This is the situation: We have 2 Asterisk servers, connected to PSTN. Phone A -- Asterisk1 -- Gateway - PSTN Phone B -- Asterisk2 -- Gateway - PSTN A dials B, call reaches B, but it rejects the call, and sends back BUSY. The CDR on B is perfect, it says BUSY. Following is from the CDR file of Asterisk2: start ,answer ,end,dur ,billable,disposiion,ama flags,, - 2/8/2005 1:06,2/8/2005 1:06, ,10,0 ,BUSY ,DOCUMENTATION But the corrosponding record in Asterisk1 says something different: start,answer ,end ,dur ,billable,disposiion,ama flags,, - 2/8/2005 1:06,2/8/2005 1:06,2/11/2005 1:51,261895,261882 ,ANSWERED ,DOCUMENTATION As you can see the billable seconds (as per the CDR) are 261882 and the call lasted for 3 days. As you can understand, this is a very serious problem. Can anyone please clarify the following? 1. Why the first Asterisk did not recognise the busy sent back from Asterisk2? 2. Has anyone faced such problems? If so how did they resolve this? 3. What precautions need to be taken to avoid this in the future? I greately appreciate any answer you can provide on this. TIA, Girish __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why should I answer a Newbie question, there thick!
It would be nice just for once to actually use a mailing list with people who are a little more sympathetic to the fact that your not a rocket scentist or molecular biologist and that you might actually need some help, without being made to feel like your completely useless and should be cleaning toilets for a living. Ahhh man not another stupid newbie question! are these people completely lazy and thick? lets postup some sarcastic comment! --- really usefull! Yes I have spent hours researching on Google, but what may take me 3 days to workout, wading through pages of out of date information, can normally be answered by some with a little experience in seconds. Opensource is about a freindly, helpfull community of people who instead of choosing the large corporate companies, decide to give the little guy a chance. Don't put people off just because their not the next Albert Einstein, otherwise the likes of Microsoft etc. have already won! -- Regards Phil Brooks Technical Support Team Brooks Computer Solutions 0115 468333 -- This message was scanned for spam and viruses by BitDefender. For more information please visit http://linux.bitdefender.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wctdm and two tdm cards
Greetings, I have a server I'm working on here with two tdm cards in it. 4 FXS and 4FX0. Both cards work fine on their own. The problem lies with using both in the system at once. I have verified the IRQ's are fine. I have tried switching the slots the cards reside in, no luck though. I am using ACPI but not APM. I am using gentoo latest, with vanilla 2.6(.10) kernel and udev. CVS as of CVS-HEAD-03/02/05-03:42:41. The problem is as follows: If I power up the system from system off, the cards both get detected If I reboot the system with reset button, ctrl alt del, or 'reboot' the TDM04P does not get detected. If I then reboot, then hit the power button, and let it turn off, then turn it back on again and boot, it detects both cards fine. I have tried searchign the list archives, but I have not had much luck. One person on IRC mentioned he's seen this before, but didn't have any solutions. Does anyone here know what might be the problem? or have a fix/work around? I know I shouldnt be rebooting servers, but I have to make sure it works upon reboot as it is going to be installed in a power-outtage happy part of the world :) TIA, Matts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940, Voicemail DTMF
Hi Craig, The only lines I seem to have in SIPDefault relating to DTMF are: # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) #dtmf_db_level: 3 dtmf_db_level: 5 dtmf_avt_payload: 101 ; Default 101 I've played with most of the settings but so far its still not working. I can still dial any phone numbers fine but I need to have another phone on my desk for voice mail :) Derek Craig Guy wrote: I set dtmfmode=inband for my 7960 in order for voicemail to work. Craig - Original Message - From: Mark Johnson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 01, 2005 11:20 PM Subject: Re: [Asterisk-Users] Cisco 7940, Voicemail DTMF Derek Conniffe wrote: Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with dtmf_avt_payload: 101 setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek I have the same line in my SIPDefault.cnf and my 7940's and 60's work OK using the same POS version as you. I don't have any suggestions. Sorry. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wctdm and two tdm cards
If I reboot the system with reset button, ctrl alt del, or 'reboot' the TDM04P does not get detected. To completely reset the TDM cards before they can be reliably detected again, you may have to completely power down the machine - even to the extent of pulling out the power plug and replacing it, then booting up. Regards, Gerald. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call waiting in Australia
Has anyone had problems with Call Waiting signals causing Zap channel or bridging hangups in AU. I was on a call the other day (Zap channel to PSTN) and the call suddenly hung up on my side. I dialled the calling party and got the call again, it seems that the bridge had dropped and that the other party had not lost the connection. As soon as I got the bridging again the other party mentioned that they had had a call waiting signal immediately before I went off the air. Any one had similar experiences, or have fixes? I'm in Australia, I have the same setup, and I had the exact same thing happen twice in the space of a few minutes, just then, while calling the same person. The person who I was calling says they don't have call waiting and were disconnected from me without warning, as I was. I have disabled call waiting with my telco. I rebooted asterisk today. Personally, I've come to the conclusion that these digium cards are a bit flaky - dunno? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why should I answer a Newbie question, there thick!
On Wed, March 2, 2005 10:38, BCS Support said: It would be nice just for once to actually use a mailing list with people who are a little more sympathetic to the fact that your not a rocket scentist or molecular biologist and that you might actually need some help, without being made to feel like your completely useless and should be cleaning toilets for a living. SNIP Yes I have spent hours researching on Google, but what may take me 3 days to workout, wading through pages of out of date information, can normally be answered by some with a little experience in seconds. SNIP Ah, but the issue here is that your questions seemed to indicate you haven't even read the basic information on the site iself (i.e. the manual), as even I (just started actively looking in to Asterisk 4 or 5 days ago) was able to find the answers to the questions you asked... I do not recall the exact questions, but I do remember agreeing with the conclusion that you apparently hadn't done any actual research, based on the questions you asked... Had I had some more time on my hands at that time, I would have replied, but as I am usually very busy, and din't have the answers ready from the top of my head at that time, I didn't... I do agree that the reaction was a bit ott, but in the basis correct... Sorry! I do wish you good luck implementing *, because I am confident this has a lot of potential... God bless! (PS: Can we please play nice now? I left my flame-retardant gear at the firestation when I quit as a volunteer!) -- Francesco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-OH323 no ringing
Hello Yves, could you please describe in more detail your problem. If you Answer() the call in the dialplan it is the correct behaviour not to hear any ringinging. If this is not the case please supply more information about your setup so that we can help you. George. Yves wrote: Hello, I'm using Asterisk stable (1.0.3) with Asterisk-oh323 (0.6.5). Everything is working fine, well, except that : when a call is made from an h323 device (gnomemeeting for example), the caller does not hear any ringing at all, he suddenly hears the person who answers the phone. That can be quite disturbing for the users. Any help would be very welcome. thank you. Yves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why should I answer a Newbie question, there thick!
On Wed, 2005-03-02 at 09:38 +, BCS Support wrote: It would be nice just for once to actually use a mailing list with people who are a little more sympathetic to the fact that your not a rocket scentist or molecular biologist and that you might actually need some help, without being made to feel like your completely useless and should be cleaning toilets for a living. Yes I have spent hours researching on Google, but what may take me 3 days to workout, wading through pages of out of date information, can normally be answered by some with a little experience in seconds. As can be seen from below your questions were answered as deeply as could be from the information you supplied. As others have said before general questions get general answers. Question Incomming Lines ISDN 2 Channel From BT (yes im in the UK) (Do I need some type of ISDN Interface Card?) Answer Yes Question Extensions 10 Users require (Can I use a computer to answer and field calls?) Answer Yes Question VOIP Phone Numbers Do we need to register some type of VIOP telephone number? (are there differnt standards or are the VOIP number accessable by all?) Answer - any other answer would have required a web site on its own. Do some more reading. Question Cost What is an average Hardware cost for this type of system? Answer See price list of those who do supply this type of system. How do you work out a hardware cost from such a general spec? As you yourself said lists are to help people but _you_ need to help them help you. Bitching when they _do_ answer your general questions is not the way to get help. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium's G.729A codec problem
Hi, all, I have buy 5 Digium's G.729A codec(it just support G.729A license) When I calll with 2 SIP UA that support G.729A and G.729B, its rtp frame have some problem when softswitch with Asterisk. The voice frame have been drop, so sometime I can't hear voice. If I want to fix the problem when softswitch G.729A and G.729B codec. What source code I must to modify ? Or some people have finished the issue, Could you show me how to do? -- Jacky ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] welltech fxo sip registration
Hi all user welltech with asterisk if any asterisk and welltech user facing any problem to register all port , please contact with welltech tech , they have patch for fxo gateway for sip. I got one for my 6 port fxo . and problem solved. thanks welltech for patch - Original Message - From: Dave Cotton [EMAIL PROTECTED] To: Asterisk List asterisk-users@lists.digium.com Sent: Wednesday, March 02, 2005 2:49 AM Subject: Re: [Asterisk-Users] Why should I answer a Newbie question, therethick! On Wed, 2005-03-02 at 09:38 +, BCS Support wrote: It would be nice just for once to actually use a mailing list with people who are a little more sympathetic to the fact that your not a rocket scentist or molecular biologist and that you might actually need some help, without being made to feel like your completely useless and should be cleaning toilets for a living. Yes I have spent hours researching on Google, but what may take me 3 days to workout, wading through pages of out of date information, can normally be answered by some with a little experience in seconds. As can be seen from below your questions were answered as deeply as could be from the information you supplied. As others have said before general questions get general answers. Question Incomming Lines ISDN 2 Channel From BT (yes im in the UK) (Do I need some type of ISDN Interface Card?) Answer Yes Question Extensions 10 Users require (Can I use a computer to answer and field calls?) Answer Yes Question VOIP Phone Numbers Do we need to register some type of VIOP telephone number? (are there differnt standards or are the VOIP number accessable by all?) Answer - any other answer would have required a web site on its own. Do some more reading. Question Cost What is an average Hardware cost for this type of system? Answer See price list of those who do supply this type of system. How do you work out a hardware cost from such a general spec? As you yourself said lists are to help people but _you_ need to help them help you. Bitching when they _do_ answer your general questions is not the way to get help. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk 1.0.5
can i install this directly or do i have to install 1.0.0 and then upgrade ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] e164.org and FWD now have peering arrangement
There is now a peering arrangement between e164.org and FreeWorldDialup which means any and all subscribers on FWD are now easily able to make enum calls by prefixing their call with **164, like wise it's almost as simple to make a call to FWD by hitting 8829990fwd number This means that for those of you wanting to send/receive calls to/from FWD subscribers you can now do so, easily and without needing to stay registered to their servers... Just prefix your caller ID with **164 if you want to make it easy for people to hit redial... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help needed with installing ZAPHFC
I need some help with installing Asterisk with ZAPHFC on my Linux system. I've installed Red Hat 9 with kernal 2.4.20-8 and upgraded it to 2.4.31-9. I've donwloaded bristuff (several versions) but all fail to compile. There seems to be some includes missing in /usr/includes/linux. Although I have significant experience with other platforms, I'm new to Linux and would need someone to help me install Asterisk with ZAPHFC. I have a Pentium III 700 Mhz system with 768 Mb RAM. Additional hardware includes a X100P, Ethernet Card and ISDN (EuroISDN, HFC) adapter. The set-up that I have in mind connects the X100P to the PSTN (analog) and the ISDN card to a ISDN PBX phone that I have at home. My initial set-up would route calls from the ISDN PBX to the analog PSTN via Asterisk, later to be extended by connecting it FWD. I'd appreciate any help in getting ZAPHFC compiled on my system. I'm not in love with Red Hat, so any working set-up is fine. The bootable Asterisks CD's don't work for me as my system contains other operating systems that I want to retain for the moment. I apologise if this question has been asked before (I couldn't find it). Alexander ___ $0 Web Hosting with up to 120MB web space, 1000 MB Transfer 10 Personalized POP and Web E-mail Accounts, and much more. Signup at www.doteasy.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.5
Am Mittwoch 02 März 2005 12:54 schrieb Kanishka Somaratne: can i install this directly or do i have to install 1.0.0 and then upgrade ? It's a complete package no upgrade needed Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.0.5
Kanishka, You can install it directly. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ Kanishka Somaratne wrote: can i install this directly or do i have to install 1.0.0 and then upgrade ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incorrect CDRs
Parts of your dialplan would be helpful to figure out what you are doing Jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting Asterisks via SIP
OK, I have installed version from CVS (version:
CVS-HEAD-03/02/05-09:33:13) and it helped. I'm able to make calls from
PBX1 to PBX2 *xor* PBX2 to PBX1, but I'm not albe to join the
configurations (to both PBX1 - PBX2 and PBX2 - PBX1). If I add peer
for other side I get fallowing error:
--
*CLI Mar 2 12:38:16 WARNING[10786]: chan_sip.c:7554 handle_response:
Forbidden - wrong password on authentication for INVITE to 'asterisk
sip:[EMAIL PROTECTED];tag=as57c8a343'
-- SIP/207-204-1764 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Got SIP response 481 Call Leg Does Not Exist back from 10.1.3.204
== Auto fallthrough, channel 'OSS/dsp' status is 'CONGESTION'
and on the other side:
*CLI Mar 2 12:38:41 NOTICE[21933]: chan_sip.c:8011 handle_request:
Failed to authenticate user asterisk
sip:[EMAIL PROTECTED];tag=as57c8a343
Below is the configuraton. The strange thing is that if I remove
[204-207] on PBX2 I'm able to call from PBX2 to PBX1. Alternatively if I
remove [207-204] from PBX1 I'm able to call from PBX2 to PBX1. If all
sections [204-207] and [207-204] are turned on I'm not able to call in
either direction.
Thank you one more time for help!
Marcin Okraszewski
=== CONFIGURATION =
PBX1 (10.1.3.207)
==
sip.conf
--
[207-204]
type=peer
username=207-204
secret=207-204
host=10.1.3.204
[204-207]
type=user
secret=204-207
extensions.conf
exten = 113,1, Dial(SIP/adamo,10,t)
exten = 158,1, Dial(SIP/okrasz,10,t)
exten = _2XX,1, Dial(SIP/207-204/${EXTEN})
PBX2 (10.1.3.204)
==
sip.conf
--
[207-204]
type=user
secret=207-204
[204-207]
type=peer
username=204-207
secret=204-207
host=10.1.3.207
extensions.conf
exten = 213,1, Dial(SIP/adamo2,10,t)
exten = 258,1, Dial(SIP/okrasz2,10,t)
exten = _1XX,1, Dial(SIP/204-207/${EXTEN})
=== END CONFIGURATION
Marcin Okraszewszki wrote:
exten = _1XX,1, Dial(SIP/pbx2:[EMAIL PROTECTED]/${EXTEN},30,r)
This syntax does not work. The extension part was just recently fixed
in CVS HEAD, but you cannot specify the secret in the dial string.
You will need to create a SIP peer for this server that contains the
IP address and secret, then you can use:
Dial(SIP/pbx2/${EXTEN})
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Re: [Asterisk-Users] [OT - somewhat] chan_sccp status
On Tue, Mar 01, 2005 at 03:50:16PM -0600, Chris Wade arranged a set of bits into the following: I hate to re-post like this, but I still haven't been able to get ahold of the chan_sccp developers (short of opening a bug report on their mantis installation just to get their attention :). I originally sent this email back at the beginning of February. I think I sent a response to this message when it was first sent. I would love to see an update as to the status of chan_sccp. Also, I'm very interest in contributing to the efforts of chan_sccp, so please, if anyone from the dev team is reading this, please drop me a line. Status: For myself, I work on what intrests me when I can (I'm now a full-time student and work 2.5 days a week). I'm slowly commiting my fixes for various things, but my three additional features (proper contention beeps, the voicemail button and better hold support) are waiting until I can get more models of phones to test against (I'm missing a 7910, a 7905/12, a 7960 (my 7940 should arrive tomorrow) and a 7920), as I posted before, SCCP is not a well defined protocol and the phones change it seemingly on a whim so it's much harder then trying to implment a standard like SIP or IAX. [Also useful are: 7935/6, 7970, 7914]. And again, if anyone has a callmanager installation tcpdump format ethernet dumps of features/phones that chan_sccp doesn't yet support are helpful (just ask before sending even 1MB of dump). I've also contacted Cisco who claimed that they don't HAVE protocol docs for SCCP (even though I have the ISBN...) and arn't willing to help out with info at all. And in regards to the New release around 20/1/05 I don't know either, and if I had admin rights on sf.net I would have long removed it, but my own e-mail's to Jan have gone unreplied. Thanks, Julien chan_sccp developer pgptBBIxNhN3V.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] where is voice conduits
According to http://www.itu.int/ITU-T/inr/forms/files/Applications-E-164.pdf Page 3 the 882 99 has been assigned to Telenor (http://www.telenor.com). So e164.org may have a problem with that prefix, if the 882 99 is ever used by Telenor. Regards, Marc ross jones wrote: on 2/28/05 09:49, Andrew Thompson at [EMAIL PROTECTED] wrote: There was a thread a month or two ago on here about voiceconduits. The general gist was they are not yet open for public business. Are there any voice conduits customers out there? if not, maybe I ought to just walk away. -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] where is voice conduits
Sorry for replying into the wrong thread. Regards, Marc Marc Storck wrote: According to http://www.itu.int/ITU-T/inr/forms/files/Applications-E-164.pdf Page 3 the 882 99 has been assigned to Telenor (http://www.telenor.com). So e164.org may have a problem with that prefix, if the 882 99 is ever used by Telenor. Regards, Marc ross jones wrote: on 2/28/05 09:49, Andrew Thompson at [EMAIL PROTECTED] wrote: There was a thread a month or two ago on here about voiceconduits. The general gist was they are not yet open for public business. Are there any voice conduits customers out there? if not, maybe I ought to just walk away. -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wctdm and two tdm cards
On March 2, 2005 05:07 am, Begumisa Gerald M wrote: To completely reset the TDM cards before they can be reliably detected again, you may have to completely power down the machine - even to the extent of pulling out the power plug and replacing it, then booting up. There's *got* to be a way to programmatically do this in the driver init; I mean there aren't many (any?) other cards (network, video, sound, etc.) cards out there that require this. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why should I answer a Newbie question, there thick!
On March 2, 2005 04:38 am, BCS Support wrote: Ahhh man not another stupid newbie question! are these people completely lazy and thick? lets postup some sarcastic comment! --- really usefull! The only time I see these types of responses is when it's clear that the question asker has not done any basic research. Yes I have spent hours researching on Google, but what may take me 3 days to workout, wading through pages of out of date information, can normally be answered by some with a little experience in seconds. Perhaps you're not correctly asking questions then. http://www.catb.org/~esr/faqs/smart-questions.html Opensource is about a freindly, helpfull community of people who instead of choosing the large corporate companies, decide to give the little guy a chance. Open source is *also* about helping yourself and not expecting a free ride. As I said earlier, the only time I see newbies flamed out here is when it's obvious that they are not doing any of their own research and want someone to do their homework for them. I gladly do that, but I always attach a rate sheet for my consulting. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 x g729 x Unable to create/find channel
Guy Decarpentrie wrote: Try to configure your Cisco type=friend in your sip.conf It is already type=friend [1234] type=friend username=1234 auth=md5 secret=supersecret deny=0.0.0.0/0.0.0.0 permit=my_ip/255.255.255.255 canreinvite=no reinvite=no host=dynamic dtmfmode=rfc2833 qualify=1800 mailbox=1234 disallow=all allow=g729 nat=yes context=cisco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Manager API - multi Originate calls
Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read inplaces that you use "originate" command and wait for an event back, does that mean you cannot place another "originate" until the event comes back ? Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) All help would be welcome - thanks Stephen Owen sip:[EMAIL PROTECTED]IM:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More NAT questions
Hi, all Still trying to get NAT working. I have following setup: PHONE 1 -- * BOX | NAT/Firewall | | NAT/Firewall | | PHONE 2 Firewall next to phone 2 has all ports open. Firewall next to Asterisk has open ports 5060 and 1:2. All of those are forwarded to Asterisk box. Both phones succesfully register with Asterisk. (I had to add NAT=yes to configuration of PHONE 2 in sip.conf to get this far). Now, problems: I can place a call from PHONE2 to PHONE1, but sound path is not established. Calls from PHONE1 to PHONE2 can not be placed at all. (I assume that this is because port 5060 is not forwarded to the phone at NAT/Firewall, but more on it later). Looking at SIP debug info, Asterisk tries to use local address of PHONE2 instead of its public IP. As a result, no info can be sent to it. I have tried to install SIPROXD on the NAT/Firewall close to Asterisk box, but this did not help. Now, we have tried to use one of the commercial VoIP service at PHONE2 location. We had to use their phone and it worked just fine without any alterations to NAT/Firewall device. I am pretty sure that they use SIP, so they did resolve the problem somehow. Sorry, there is no technical info available on this service. Did anyone succeeded in doing this setup? I know, IAX is a better way, but I can not setup many Asterisk boxes. Basically, I am doing it for a friend. He is working for a small medical company. They have number of offices that are not open every day and offices are too small to put Asterisk box in each one. There will be 1-3 IP phones in each office, except central one. Central one will need Asterisk, the rest should be on their own. Any help is greatly appreciated. Thanks, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wctdm and two tdm cards
I have a server I'm working on here with two tdm cards in it. 4 FXS and 4FX0. Both cards work fine on their own. The problem lies with using both in the system at once. I have verified the IRQ's are fine. I have tried switching the slots the cards reside in, no luck though. I am using ACPI but not APM. I am using gentoo latest, with vanilla 2.6(.10) kernel and udev. CVS as of CVS-HEAD-03/02/05-03:42:41. The problem is as follows: If I power up the system from system off, the cards both get detected If I reboot the system with reset button, ctrl alt del, or 'reboot' the TDM04P does not get detected. If I then reboot, then hit the power button, and let it turn off, then turn it back on again and boot, it detects both cards fine. I have tried searchign the list archives, but I have not had much luck. One person on IRC mentioned he's seen this before, but didn't have any solutions. Does anyone here know what might be the problem? or have a fix/work around? I know I shouldnt be rebooting servers, but I have to make sure it works upon reboot as it is going to be installed in a power-outtage happy part of the world :) I'm not having any problems like that with RHv9 (2.4 kernel), so I'd have to guess the issue is 'timing' related in whatever script that loads your tdm-zaptel drivers. As I recall (as a non-v2.6 user), there was an issue with timing and someone added a sleep/wait statement in the startup script to bypass the problem. Might consider finding your startup and add some additional time to that sleep/wait. Another approach to isolating the problem is to load the drivers by hand paying close attention to error messages, delays, etc. If your not sure how to do that, read your startup script and simply do those steps manually. Someone mentioned unplugging power and/or removing the card. That approach is totally BS. The same startup process is run regardless of whether one is rebooting or starting from power-on. There is nothing on the tdm card that stores values (no flash, no battery backup mem, etc). If the startup script operates one time from any startup mode, it is setting the tdm registers, etc, correctly. I'm away from the office this week, but I recall there was a readme shipped with the zaptel source that discusses kernel 2.6 timing issues. Might look for that in your src directory. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dual X100P cards
I know the X100P cards are not supported by Digium any more, but for home office use, are they still acceptable? I have two POTS lines, one residential and one business line comming into the house. I'd like to get both into my * server and $15 total compared to $100 for the newer TDMxxx card sure is desirable. Having said that, will the sound quality, functionality, stability, etc. be as good? I don't want to spend any more than I have to, but if the X100P cards are crappy, unstable, whatever than saving a couple bucks is not worth it. - Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cvs stable and 1.0.5
I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would always get the newest releases. However, I just did a fresh update and install from cvs stable and it reports as only being v1.0.3. Should I just be using the tarballs rather than the cvs -r 1_0? Or maybe my initial cvs was incorrect? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323_OUTCODEC Unsupported
Hello Karim,
do you have properly configured g723 in your oh323.conf ?
you must have :
codec=G7231
in your [codec] section.
George.
M. Ehsanul Karim wrote:
Whenever I do :
SetGlobalVar(OH323_OUTCODEC=g723.1)
I get this meesage
Unsupported ${OH323_OUTCODEC} value (g723.1)!
g723.1 works fien for SIP and if I put
SetGlobalVar(OH323_OUTCODEC=g729). It works fine as well.
I have enabled all g723 in oh323.conf.
Any sugegstions ?
Ehsanul Karim
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RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls
Hello, You can do either, you can send multiple Originate actions in a long line without waiting for a response back(although the responses do usually come back very fast) or you can open multiple connections using each one to Originate a new call. We use the multiple connection method in the astGUIclient suite because if you get a pause or lag in Manager output on a single connection(which does happen) it will hold up all of the Actions you are trying to send after it. Take a look at the ACQS(Asterisk Central Queue System) part of the astGUIclient suite. It allows you to queue up Actions in a database and the server will send the actions to the asterisk server almost immediately. We've been using this for quite a while now and it is very reliable. MATT--- -Original Message- From: Stephen Owen hosted [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 02, 2005 7:28 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Manager API - multi Originate calls Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read in places that you use originate command and wait for an event back, does that mean you cannot place another originate until the event comes back ? Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) All help would be welcome - thanks Stephen Owen sip:[EMAIL PROTECTED] IM:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callback on busy
consider this scenario: A Calls B B transfers A to C C (is busy or does not answer) so A backs to B On Tue, 1 Mar 2005 23:07:17 +0330, Paradise Dove [EMAIL PROTECTED] wrote: consider this scenario: A Calls B B transfers A to C C (is busy or does not answer) so B backs to A On Tue, 1 Mar 2005 14:25:33 -0500, C F [EMAIL PROTECTED] wrote: use retrydial. in the cli type show application retrydial have fun. On Tue, 1 Mar 2005 22:17:35 +0330, Paradise Dove [EMAIL PROTECTED] wrote: hi, is there anyway to implement callback on busy and callback on no answer on asterisk? has anybody done this before? thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls
your a star thanks - Original Message - From: mattf [EMAIL PROTECTED] To: 'Stephen Owen hosted' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, March 02, 2005 12:50 PM Subject: RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls Hello, You can do either, you can send multiple Originate actions in a long line without waiting for a response back(although the responses do usually come back very fast) or you can open multiple connections using each one to Originate a new call. We use the multiple connection method in the astGUIclient suite because if you get a pause or lag in Manager output on a single connection(which does happen) it will hold up all of the Actions you are trying to send after it. Take a look at the ACQS(Asterisk Central Queue System) part of the astGUIclient suite. It allows you to queue up Actions in a database and the server will send the actions to the asterisk server almost immediately. We've been using this for quite a while now and it is very reliable. MATT--- -Original Message- From: Stephen Owen hosted [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 02, 2005 7:28 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Manager API - multi Originate calls Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read in places that you use originate command and wait for an event back, does that mean you cannot place another originate until the event comes back ? Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) All help would be welcome - thanks Stephen Owen sip:[EMAIL PROTECTED] IM:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Hello, You can do either, you can send multiple Originate actions in a long line without waiting for a response back(although the responses do usually come back very fast) or you can open multiple connections using each one to Originate a new call. We use the multiple connection method in the astGUIclient suite because if you get a pause or lag in Manager output on a single connection(which does happen) it will hold up all of the Actions you are trying to send after it. [...] Content analysis details: (0.1 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dual X100P cards
We are having good luck with these cards up to 4 cards in a server. However make sure they do not share IRQ's or you will get weird noises on the line. Anything over 4 lines and we use a TDM card. We also do a six line server that is a 4 line TDM and 2 X100P's and that is working great also. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary MacKay Sent: Wednesday, March 02, 2005 6:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Dual X100P cards I know the X100P cards are not supported by Digium any more, but for home office use, are they still acceptable? I have two POTS lines, one residential and one business line comming into the house. I'd like to get both into my * server and $15 total compared to $100 for the newer TDMxxx card sure is desirable. Having said that, will the sound quality, functionality, stability, etc. be as good? I don't want to spend any more than I have to, but if the X100P cards are crappy, unstable, whatever than saving a couple bucks is not worth it. - Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More NAT questions
Still trying to get NAT working. I have following setup: PHONE 1 -- * BOX | NAT/Firewall | | NAT/Firewall | | PHONE 2 Firewall next to phone 2 has all ports open. Firewall next to Asterisk has open ports 5060 and 1:2. All of those are forwarded to Asterisk box. Both phones succesfully register with Asterisk. (I had to add NAT=yes to configuration of PHONE 2 in sip.conf to get this far). Now, problems: I can place a call from PHONE2 to PHONE1, but sound path is not established. Calls from PHONE1 to PHONE2 can not be placed at all. (I assume that this is because port 5060 is not forwarded to the phone at NAT/Firewall, but more on it later). Looking at SIP debug info, Asterisk tries to use local address of PHONE2 instead of its public IP. As a result, no info can be sent to it. I have tried to install SIPROXD on the NAT/Firewall close to Asterisk box, but this did not help. Now, we have tried to use one of the commercial VoIP service at PHONE2 location. We had to use their phone and it worked just fine without any alterations to NAT/Firewall device. I am pretty sure that they use SIP, so they did resolve the problem somehow. Sorry, there is no technical info available on this service. Did anyone succeeded in doing this setup? I know, IAX is a better way, but I can not setup many Asterisk boxes. Basically, I am doing it for a friend. He is working for a small medical company. They have number of offices that are not open every day and offices are too small to put Asterisk box in each one. There will be 1-3 IP phones in each office, except central one. Central one will need Asterisk, the rest should be on their own. As you have already noted, trying to implement this with two nat boxes is very difficult and in some cases impossible. The only way to know for sure what is happening is to use a packet analyzer (eg, ethereal) to observe the packets on the inside and outside of each nat box. Keep in mind that no all nat boxes operate the same way; there are major differences even though we tend to characterize nat boxes as all the same. The rtp ports used for voice (1:2 in your example) vary by phone type. Cisco uses a different range of ports, Xten another range, Grandsteam yet another. The ports you have listed are what asterisk uses and are probably not the same ports as what your remote phones use. Therefore, the exact ports that you need to open are dependent upon exactly which phones you deploy, and on well you understand the handshaking that goes on end-to-end when establishing a sip call. Likewise, not all phones operate the same from behind a nat box. The snom phones happen to be very good in terms of discovering where it sits in the end-to-end picture, while other phones are either very poor or don't handle nat well at all. Since you didn't mention what type of phones you use, there's no way to guess at what might be happening. Even if you post the phone type, its not going to be of much use to the rest of us since we don't know the type of nat box in use. You also might find (later) that not all nat boxes support multiple phones behind a nat box. Eg, if one phone is made to work and its in use, the second phone behind that nat box will probably fail. Some folks have been successful with multiple phones while many others have not, and most do not know why. You might be able to discover the nat problems by tracing packets (with ethereal) from inside and outside that asterisk nat box, but I'd have to guess you'll have less then a 50% chance of seeing the issues without traces from inside the nat box at the phone location also. You really need a clear understanding of the exact IP addresses and port numbers from each location to know how to solve the problem. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Manager API - multi Originate calls
read inplaces that you use originate command and wait for an event back, does that mean you cannot place another originate until the event comes back ? Not in my experience. Originate will not send an event to the caller until either the intended caller (that is the extension used in Originate) has picked up their phone or a timeout occurs because the intended caller does not pick up their phone. You can send as many originate requests as you like but they will fail if more than one uses the same extension at the same time. The issue you will face is determining which event generated by Asterisk belongs to which origination request. For this reason, the Manager API allows you to specify an ActionID on any command. An ActionID is any string of characters that you use to uniquely identify each command use issue. Asterisk will include the ActionID with each related event so you know which events to respond to and which to ignore. You will see many events generated by Asterisk only some of which will relate to your command. The others will be events that Asterisk raises (for example when a phone registers) or events in response to commands issues by other Manager API users and at the command line. Take a look at Nicolas Gudinos Flash Operator Panel (www.asternic.org) as it used the manager API extensively (albeit through a proxy) and will typically make many requests via the Manger API. Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) Again, not in my experience. Lyquidity Solutions Limited +44 (0) 208 241 0500 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Owen hosted Sent: March 02, 2005 12:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Manager API - multi Originate calls Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read inplaces that you use originate command and wait for an event back, does that mean you cannot place another originate until the event comes back ? Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) All help would be welcome - thanks Stephen Owen sip:[EMAIL PROTECTED] IM:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual X100P cards
I know the X100P cards are not supported by Digium any more, but for home office use, are they still acceptable? I have two POTS lines, one residential and one business line comming into the house. I'd like to get both into my * server and $15 total compared to $100 for the newer TDMxxx card sure is desirable. Having said that, will the sound quality, functionality, stability, etc. be as good? I don't want to spend any more than I have to, but if the X100P cards are crappy, unstable, whatever than saving a couple bucks is not worth it. There are lots of folks that have implemented dual cards and are working reasonably well. I've had two working and since have replaced them with a tdm card. If you are in the US, the cards can be made to function but you'll need to pay attention to the motherboard in use, shared interrupts, etc, etc. For the small difference in expenditures, I'd seriously consider a pair of spa3000's or something like that. They will give you far more flexibility and a lot less support problems. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on MS Virtual Server
Hello I downloaded Astwind and get working the network (means can access to Internet through MS Windows). DEbian and Asterisk files are updated from Internet. But When I make install in Zaptel (it was my first make) I got many errors. Acoording to one manual this happens when we do not have modeversion .h kernel header file (according to it, it should reside in /usr/src/linux) which in /usr/src/linux, a make menuconfig will create it. BuT I do not have the linux dir (in /usr/src) and kernel source files thus modversion.h file. In addition I do not know how to download kernel files to linux directory (I tried apt-get but I could not format properly the /etc/spt/source.list file) Could you help. Am I in the correct path? Regards Turgut -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Sunday, January 30, 2005 8:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk on MS Virtual Server On Sun, 30 Jan 2005, Paul Tyreman wrote: http://www.digium.com/index.php?menu=astwind I think this may be worth a look, I'm downloading it as I type this e-mail... I didn't know Asterisk had the possibility of being run on a windows machine and while it's not as stable as a Linux implementation, it might just do for the moment, as I don't have many users. Is there any documentation on this windows based software, or if not, do you know where I can get more info on it ? Thanks, Paul. Paul, Since I maintain AstWind, please feel free to give me your feedback when you get a chance. I'm working on re-packaging and updating AstWind so that it contains a 2.6.8 Colinux kernel, 1.0.5 Asterisk and all of the latest Debian Updates, with a real installer. I.E. not a crappy batch file that just copies stuff over! ;) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323 bugtracker
Hello all, In an attempt to make easier and more effective the management of the various issues/features/bugs of asterisk-oh323, I have setup a bugtracker at: https://skylab.inaccessnetworks.com/mantis Please direct all the bug reports and contributed patches there. Thanks, Michael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why should I answer a Newbie question, there thick!
If you are only new to asterisk go and download [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ It's a iso you can download that does all of the configuring and setup for you automatically. You can be up and running n only 30 mins (even I managed it). Cheers dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BCS Support Sent: Wednesday, March 02, 2005 4:38 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Why should I answer a Newbie question, there thick! It would be nice just for once to actually use a mailing list with people who are a little more sympathetic to the fact that your not a rocket scentist or molecular biologist and that you might actually need some help, without being made to feel like your completely useless and should be cleaning toilets for a living. Ahhh man not another stupid newbie question! are these people completely lazy and thick? lets postup some sarcastic comment! --- really usefull! Yes I have spent hours researching on Google, but what may take me 3 days to workout, wading through pages of out of date information, can normally be answered by some with a little experience in seconds. Opensource is about a freindly, helpfull community of people who instead of choosing the large corporate companies, decide to give the little guy a chance. Don't put people off just because their not the next Albert Einstein, otherwise the likes of Microsoft etc. have already won! -- Regards Phil Brooks Technical Support Team Brooks Computer Solutions 0115 468333 -- This message was scanned for spam and viruses by BitDefender. For more information please visit http://linux.bitdefender.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wctdm and two tdm cards
On March 2, 2005 07:26 am, Rich Adamson wrote: I'm not having any problems like that with RHv9 (2.4 kernel), so I'd have to guess the issue is 'timing' related in whatever script that loads your tdm-zaptel drivers. I disagree. My TDM430P will misdetect/miss modules entirely. rmmod and modprobe wctdm (not zaptel, it stays loaded) and it works. rmmod and reload again and it's missing/misdetecting modules. It's a driver bug. I haven't mantis'd it only because it hasn't caused me any significant grief and I don't have a testcase that will reliably work. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting Asterisks via SIP
OK, finally I made it working. And it works also with version 1.0.5. The
configuration:
PBX1 (10.1.3.207)
==
sip.conf
--
[pbx]
type=friend
username=pbx
secret=pbx
host=10.1.3.204
extensions.conf
exten = 113,1, Dial(SIP/adamo,10,t)
exten = 158,1, Dial(SIP/okrasz,10,t)
exten = _2XX,1, Dial(SIP/pbx/${EXTEN})
PBX2 (10.1.3.204)
==
sip.conf
--
[pbx]
type=friend
username=pbx
secret=pbx
host=10.1.3.207
extensions.conf
exten = 213,1, Dial(SIP/adamo2,10,t)
exten = 258,1, Dial(SIP/okrasz2,10,t)
exten = _1XX,1, Dial(SIP/pbx/${EXTEN})
Maybe someone will find if he needs it :)
Regards
Marcin Okraszewski
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RE: [Asterisk-Users] cvs stable and 1.0.5
Are you sure you're not looking at the date? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Wednesday, March 02, 2005 7:47 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cvs stable and 1.0.5 I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would always get the newest releases. However, I just did a fresh update and install from cvs stable and it reports as only being v1.0.3. Should I just be using the tarballs rather than the cvs -r 1_0? Or maybe my initial cvs was incorrect? Thanks! -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cvs stable and 1.0.5
On Wed, 2 Mar 2005 07:46:33 -0500, Michael George [EMAIL PROTECTED] wrote: I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would always get the newest releases. However, I just did a fresh update and install from cvs stable and it reports as only being v1.0.3. Actually, 1.0.6 is out... Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi - fax patch - crash
WARNING[pid]: CAPI[contr3/123456]/178 already has PBX structure?? WARNING[pid]: CAPI[contr3/123456]/178 already has a call record?? WARNING[pid]: CDR on channel 'CAPI[contr3/12345]/177' already started WARNING[pid]: Thread 1109916592 Blocking 'CAPI[contr3/123456]/178', already blocked by thread 1116277680 in procedure ast_waitfor_nandfds WARNING[pid]: Stack is not at expected value WARNING[pid]: Stack returned to an unexpected place! WARNING[pid]: Stack is not at expected value WARNING[pid]: Stack returned to an unexpected place! It is Asterisk 1.0.5, chan_capi 0.3.5, http://200.59.203.76/pub/chan_capi-0.3.5-patch.stable.diff 123456 is the extension of some SIP phone. Unfortunately there is no further information available and I do not know how to reproduce the crash. Any ideas? -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MozPhone
Glenn A. Thompson a écrit : Hi, Is anyone using mozPhone? If so any feedback you can provide? Yes. For what I'm doing with it work. Could be improved. -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] /dev/zap not created
I installed asterisk on Fedora Core 2 kernel 2.6.5. I followed the standard procedure. zaptel-libpri-asterisk. The thing is that I constantly get the error message: line 4: Unable to open master device '/dev/zap/ctl' where the file zaptel.conf contains only 4 files: fxoks=1 fxsks=4 defaultzone=us loadzone=us I cant run asterisk and get a load of error messages. When I tried to check the directory /dev/zap, it wasnt there. It isnt created during installation. Can someone help me out. Apart from that, should I get a dialtone in the fxs module when the tdm400p is connected in the computer? because I have no way of checking whether my device is correctly connected. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple lines
Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on MS Virtual Server
Turgut Abacioglu wrote: Hello I downloaded Astwind and get working the network (means can access to Internet through MS Windows). DEbian and Asterisk files are updated from Internet. But When I make install in Zaptel (it was my first make) I got many errors. Acoording to one manual this happens when we do not have modeversion .h kernel header file (according to it, it should reside in /usr/src/linux) which in /usr/src/linux, a make menuconfig will create it. BuT I do not have the linux dir (in /usr/src) and kernel source files thus modversion.h file. In addition I do not know how to download kernel files to linux directory (I tried apt-get but I could not format properly the /etc/spt/source.list file) Could you help. Am I in the correct path? No, you are not. Zaptel is a driver to hardware cards. CoLinux (on which Astwind is based) is a virtual Linux running as a Windows task. Virtual here means - no hardware. In short, you can install or otherwise use any hardware cards, like Zaptel, with Asterisk when running on CoLinux and generally, I'll advise you to not use Astwind for anything other then playing. It's a nice toy, but that is all. Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] More NAT questions
Still trying to get NAT working. Try adding a canreinvite=no. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 Inbound Dialing Problem
dhananjay sarnaik wrote: Thanks for the information. But still we are facing the same problem. We tried upgrading the firmware to latest available on sipura website and still the result is same. Does any specific DTMF setting required? we have tried all the 3 options in asterisk (inband, rfc2833 and info) but no luck In your SIP.conf make sure it's INBAND and the INBAND is specified on the SIPURA 3000. I had the same problem and that solved it. Joe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why should I answer a Newbie question, there thick!
On Mar 2, 2005, at 4:38 AM, BCS Support wrote: [snip] Opensource is about a freindly, helpfull community of people who instead of choosing the large corporate companies, decide to give the little guy a chance. Don't put people off just because their not the next Albert Einstein, otherwise the likes of Microsoft etc. have already won! [snip] I beg to differ. Open Source is not and should not be anti-corporate. Open Source is simply a means to create **better** software, not about communism and looking out for the little guy. Don't forget that * is sponsored by a Big Greedy Company. Don't use Open Source as a social statement, use it only if the software created meets your needs better. Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: dialing application - newbie question
'auto-answer' script for the 79XX phones. It basically telnets into the phone and presses the answer key Thanks Chris I suppose I could make a dial out command via telnet as well for the cisco. Other option I want to try is using agents - this to allow a degree of roaming users - course with the cisco 79xx I reckon they would have to be off hook the whole time... maybe I could using line 2 for this.. anyway worth some experimenting. :) Walt _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls
ActionID does not return in all events related to an Action sent, sometimes it will just send you a success message and nothing more. Just try Originating a call from a meetme room over an outside line. You will get about 150 lines of output and only one message will have the ActionID in it, the success message. On the other hand the callerID is placed on many more of the events in the output. It is still the case that if you do complex Manager Actions, the ONLY solution for tracking a call is to use a custom CallerID. Action: OriginateExten: 8600080Channel: local/[EMAIL PROTECTED]Context: defaultPriority: 1Callerid: DF345678901234567890Actionid: AID45678901234567890 MATT--- -Original Message-From: Bill Seddon [mailto:[EMAIL PROTECTED]Sent: Wednesday, March 02, 2005 8:06 AMTo: Stephen Owen hosted; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Asterisk Manager API - multi "Originate" calls read inplaces that you use "originate" command and wait for an event back, does that mean you cannot place another "originate" until the event comes back ? Not in my experience. Originate will not send an event to the caller until either the intended caller (that is the extension used in Originate) has picked up their phone or a timeout occurs because the intended caller does not pick up their phone. You can send as many originate requests as you like but they will fail if more than one uses the same extension at the same time. The issue you will face is determining which event generated by Asterisk belongs to which origination request. For this reason, the Manager API allows you to specify an ActionID on any command. An ActionID is any string of characters that you use to uniquely identify each command use issue. Asterisk will include the ActionID with each related event so you know which events to respond to and which to ignore. You will see many events generated by Asterisk only some of which will relate to your command. The others will be events that Asterisk raises (for example when a phone registers) or events in response to commands issues by other Manager API users and at the command line. Take a look at Nicolas Gudinos Flash Operator Panel (www.asternic.org) as it used the manager API extensively (albeit through a proxy) and will typically make many requests via the Manger API. Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) Again, not in my experience. Lyquidity Solutions Limited +44 (0) 208 241 0500 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Owen hostedSent: March 02, 2005 12:28 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk Manager API - multi "Originate" calls Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read inplaces that you use "originate" command and wait for an event back, does that mean you cannot place another "originate" until the event comes back ? Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) All help would be welcome - thanks Stephen Owen sip:[EMAIL PROTECTED]IM:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cvs stable and 1.0.5
Michael George wrote: I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would always get the newest releases. However, I just did a fresh update and install from cvs stable and it reports as only being v1.0.3. Should I just be using the tarballs rather than the cvs -r 1_0? Or maybe my initial cvs was incorrect? You forgot to rm .version in the source berfore building it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dual Asterisk Servers
Due to the unfortunate nature of Wikis, the section on voip-info.org that deals with dual asterisk servers is full of pretty bad and outdataed examples. What I'm trying to do is distribute small asterisk boxes to remote offices that have SIP clients connected inside the network, and ship any outbound calls to a central asterisk server via IAX that is in turn connected to the PSTN and some VOIP LD providers. I also want 4 digit dialing between sites, where the 1st digit is always the site id to route the call to So at the master site (site 1) we can call Siet 2 (remote site) by dialing 2XXX. Site two can call extensions at the master by dialing 1XXX etc. The relatively new switch dialplan command seems like it will assist in accomplishing this, but does anyone have a simple IAX config and dialplan that will help me understand how this all works? Regards, NIk Martin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT - somewhat] chan_sccp status
Julien Goodwin wrote: On Tue, Mar 01, 2005 at 03:50:16PM -0600, Chris Wade arranged a set of bits into the following: I hate to re-post like this, but I still haven't been able to get ahold of the chan_sccp developers (short of opening a bug report on their mantis installation just to get their attention :). I originally sent this email back at the beginning of February. I think I sent a response to this message when it was first sent. Ok, didn't see it, but no problem. I would love to see an update as to the status of chan_sccp. Also, I'm very interest in contributing to the efforts of chan_sccp, so please, if anyone from the dev team is reading this, please drop me a line. Status: For myself, I work on what intrests me when I can (I'm now a full-time student and work 2.5 days a week). I'm slowly commiting my fixes for various things, but my three additional features (proper contention beeps, the voicemail button and better hold support) are waiting until I can get more models of phones to test against (I'm missing a 7910, a 7905/12, a 7960 (my 7940 should arrive tomorrow) and a 7920), as I posted before, SCCP is not a well defined protocol and the phones change it seemingly on a whim so it's much harder then trying to implment a standard like SIP or IAX. [Also useful are: 7935/6, 7970, 7914]. And again, if anyone has a callmanager installation tcpdump format ethernet dumps of features/phones that chan_sccp doesn't yet support are helpful (just ask before sending even 1MB of dump). I see - understandable. I'll see what I can do to help. BTW, instead of waiting to have access to those phones, why not put your patches on the chan_sccp bug-tracker and allow those of us who already have those phones to test for you? I would be glad to setup a few 7940's connected to a test * server - I'll even give you ssh access to it if you need it. I'll even be your eyes/ears/fingers as needed to test those functions. My _issue_ with chan_sccp is that development in the main cvs seems to be so slow it's hard to tell what actually needs work. I would enjoy helping as much as I can - I just cannot see what needs to be done, other than the obvious stuff which I'm not sure how the 'core' developers want implemented (I'm speaking of button templates, etc. here). Basically, I don't want to work on something somebody else is already working on, I want to help with something that is 'sitting' and stagnating. I've also contacted Cisco who claimed that they don't HAVE protocol docs for SCCP (even though I have the ISBN...) and arn't willing to help out with info at all. I know the feeling :( And in regards to the New release around 20/1/05 I don't know either, and if I had admin rights on sf.net I would have long removed it, but my own e-mail's to Jan have gone unreplied. Ugh, sounds to me like the left hand doesn't know what the right is doing? Is there a roadmap for chan_sccp development? Thanks, Julien chan_sccp developer Well, now at least I have direct contact info for one of the chan_sccp devs. Thanks, Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: music on hold trouble
I too am having the same problem with =VS from last night. From my debugging, * never attempts to start MOH. Anyone else =ound this? Me too Music on hold - with SIP handsets at least - stopped working for me with asterisk 1.0.6 and cvs. If I downgraded to 1.0.5 works fine, upgrade and it stops working. all versions work fine if a dial an extension for music on hold. Cheers Walt _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE405P/zttool
Can anyone with a TE405P tell me what zttool tells them the type of card installed. With ours it says it's a TE410P. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /dev/zap not created
be sure to run service zaptel start and also make install inside zaptel one of those two commands creates all the necessary /dev stuff. -Matthew - Original Message - From: Rizwan Chaudhry [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, March 02, 2005 9:02 AM Subject: [Asterisk-Users] /dev/zap not created I installed asterisk on Fedora Core 2 kernel 2.6.5. I followed the standard procedure. zaptel-libpri-asterisk. The thing is that I constantly get the error message: line 4: Unable to open master device '/dev/zap/ctl' where the file zaptel.conf contains only 4 files: fxoks=1 fxsks=4 defaultzone=us loadzone=us I cant run asterisk and get a load of error messages. When I tried to check the directory /dev/zap, it wasnt there. It isnt created during installation. Can someone help me out. Apart from that, should I get a dialtone in the fxs module when the tdm400p is connected in the computer? because I have no way of checking whether my device is correctly connected. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to change fxo_mode
After ztcfg, /var/log/messages reads Module 3: Installed -- AUTO FXO (FCC mode) How can I change this FCC mode to something else? Soner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /dev/zap not created
Hey Rizwan, On 2 Mar 2005, at 15:02, Rizwan Chaudhry wrote: I installed asterisk on Fedora Core 2 kernel 2.6.5. I followed the standard procedure. zaptel-libpri-asterisk. The thing is that I constantly get the error message: line 4: Unable to open master device '/dev/zap/ctl' I'm not 100% sure, but I think Fedora Core 2 uses UDEV. Look through the output of ps -A and see if there is a udevd running. If there is you're running udev and need to read README.udev which is in the zaptel source directory. where the file zaptel.conf contains only 4 files: fxoks=1 fxsks=4 defaultzone=us loadzone=us I cant run asterisk and get a load of error messages. When I tried to check the directory /dev/zap, it wasnt there. It isnt created during installation. Can someone help me out. Apart from that, should I get a dialtone in the fxs module when the tdm400p is connected in the computer? because I have no way of checking whether my device is correctly connected. Things won't be properly loaded until you have udev sorted. As far as I know anyway. Phil. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cvs stable and 1.0.5
On Wed, Mar 02, 2005 at 09:49:02AM -0500, Clay Reiche wrote: Are you sure you're not looking at the date? Oh, you are probably right. It is 1-0-03/01/05, so that's 1.0 as of 3/1/5, not 1.0.3. So it appears, then, that the cvs will only display 1.0 and the .x part is only relevant for the releases. I also noticed that it's not recommended that one use the CVS version (even of stable) if not watching the asterisk-cvs list. Maybe, then, it would be best for me to revert to using the releases. What is the opinion of the list? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Wednesday, March 02, 2005 7:47 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cvs stable and 1.0.5 I see that 1.0.5 is out. I thought that if I am tracking cvs v1.0.x I would always get the newest releases. However, I just did a fresh update and install from cvs stable and it reports as only being v1.0.3. Should I just be using the tarballs rather than the cvs -r 1_0? Or maybe my initial cvs was incorrect? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple lines
David, please search the wiki for meetme rooms; this is a standard feature. If you want to be able to the control those calls from a web interface do a search for meetme2 If you are only new to asterisk go and download [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ It's a iso you can download that does all of the configuring and setup for you automatically. Cheers dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Masure Sent: Wednesday, March 02, 2005 9:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Multiple lines Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why should I answer a Newbie question, there thick!
On Wed, 2005-03-02 at 09:38 +, BCS Support wrote: It would be nice just for once to actually use a mailing list with people who are a little more sympathetic to the fact that your not a rocket scentist or molecular biologist and that you might actually need some help, without being made to feel like your completely useless and should be cleaning toilets for a living. But someone has to clean toilets. Microsoft isn't the only one to blame for the reported millions of infected computers on the internet. Some people really shouldn't be tasked with administration of a machine let alone the provisioning of a phone system. Ahhh man not another stupid newbie question! are these people completely lazy and thick? lets postup some sarcastic comment! --- really usefull! Yes I have spent hours researching on Google, but what may take me 3 days to workout, wading through pages of out of date information, can normally be answered by some with a little experience in seconds. Sounds like either you haven't made it up the first part of the learning curve and/or your search/research skills still need work. The latter part is not within the scope of this mailing list. If you can do basic research, you should be able to then ask direct questions with the proper amount of details that it becomes an interesting question to those who you would want to answer your question. Opensource is about a freindly, helpfull community of people who instead of choosing the large corporate companies, decide to give the little guy a chance. As others have pointed out, this is not true. No where is friendly a requirement. No where is helpful a requirement. No where is opensource a anti corporate tool. Examples, Go look up the reputations of some of the biggest names in opensource software. ESR went to the trouble to write a fairly long paper just so you could learn how not to get quickly flamed even by himself. There are some well known BSD developers that are well known for their treatment of people who don't agree with them let alone just act like newbies. Don't put people off just because their not the next Albert Einstein, otherwise the likes of Microsoft etc. have already won! Microsoft did win, we are just about to take the trophy away. Maybe you should use that whiny line on your next employer and see if it helps you get the job. Of course you might find that whiny adults actually generate a repulsive reaction by other adults and you are actually more likely to receive more of the same treatment. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why should I answer a Newbie question, therethick!
Arrgh, Why should I answer a Newbie question, they are thick! Why is it so difficult to just ignore any question with Newbie in it? Everyone has to start somewhere. At least the newbie found the list. The worse you can do is kick sand in their face. No newbie's means no new customers or developers who might be able to contribute. This list is owned by no one. When someone drives off the road into a ditch in a snow storm they last thing they need is someone telling them they should have invested in snow chains and defensive driving lessons before leaving the house. Newbies need help getting out of the ditch so traffic can continue to flow and the rubber neckers can be abated. If you are not willing to pull off to the side of the road and help the fool by pushing their car out of the ditch you have no right to give him the finger as you drive past. Race The Tryant Vanderdecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple lines
Dean, Thank you for your answer but fromwhat I know meetme is able to solde the conference problem, but how can I for example receive 2 phone calls at the same time on 1 phone and just switching from one line to another ? In my current config, I make a phone call and the SIP phone is answering, when trying to make a second call, I've got the music to hold me till first conversation has ended. Meanwhile, the sip phone user doesn't know there is a call waiting and so, he won't answer the line Is there a solution to that problem ? Thanks David -Message d'origine-De: dean collins [mailto:[EMAIL PROTECTED]Envoyé: mercredi 2 mars 2005 17:02À: Asterisk Users Mailing List - Non-Commercial DiscussionObjet: RE: [Asterisk-Users] Multiple lines David, please search the wiki for meetme rooms; this is a standard feature. If you want to be able to the control those calls from a web interface do a search for meetme2 If you are only new to asterisk go and download [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ It's a iso you can download that does all of the configuring and setup for you automatically. Cheers dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David MasureSent: Wednesday, March 02, 2005 9:58 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Multiple lines Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dual Asterisk Servers
On Wed, 2005-03-02 at 09:22 -0600, Nik Martin wrote: The relatively new switch dialplan command seems like it will assist in accomplishing this, but does anyone have a simple IAX config and dialplan that will help me understand how this all works? switch is OLD. I have been using switch now for over 3 years. switch is easy, you just include it in the appropriate context and it will contact the remote machines for dialplan completion. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Park Craches asterisk
So actually a problem with the binary Version for Debian (bristuffed). beacuse I have made a clean install from CVS and everything is ok. But I have other two asterisk from CVS which have this problem but there are not Bristuffed. May be it is problem when we have za Zap device ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to change fxo_mode
Sorry for littering the maillist, I've found it myself, I've changed the wctdm.c file and make install'ed zaptel drivers, now it shows: Module 3: Installed -- AUTO FXO (TURKEY mode) But I am not sure if this is the best way. And if my mode settings (in fxo_mode struct) are not correct, what kind of problems would I face? Or should this e-mail be sent to the developers' list? - Original Message - From: Soner Tari [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 02, 2005 5:39 PM Subject: [Asterisk-Users] How to change fxo_mode After ztcfg, /var/log/messages reads Module 3: Installed -- AUTO FXO (FCC mode) How can I change this FCC mode to something else? Soner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why should I answer a Newbie question, therethick!
On Wed, 2005-03-02 at 11:01 -0500, Race Vanderdecken wrote: This list is owned by no one. Actually it is owned by Digium. It has many contributers though. When someone drives off the road into a ditch in a snow storm they last thing they need is someone telling them they should have invested in snow chains and defensive driving lessons before leaving the house. Newbies need help getting out of the ditch so traffic can continue to flow and the rubber neckers can be abated. If you are not willing to pull off to the side of the road and help the fool by pushing their car out of the ditch you have no right to give him the finger as you drive past. As a person who spent 9 hours in traffic last winter just to drive 15 miles due to idiots who should have just stayed home, I think your analogy breaks down. At some point, you either need to learn to drive or you pay someone else to transport you or your stuff. Same applies to computer work, either you can do the work yourself or you pay someone else to do it. Even your snow driver analogy works here, you either get yourself out of the ditch or you pay someone to do it for you. The payment is not always monetary. Sometimes the payment is just a showing of sufficient effort. Back to your snow driver analogy, if the driver in the ditch is just waiting in the car for you to come over and push them out without even attempting anything on their own, you would be less inclined to bother. You would be even less inclined to continue exerting your own effort if the driver was not cooperating or wasn't even interested in getting out to help push. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Administration manual for Sipura-841?
Have you seen the user guide? http://www.sipura.com/Documents/SPA841UserGuide.pdf Yes, and it's actually not bad, though a bit wordy. There's almost not information on actually programming the device though. I've been trying to understand how all the configuration settings about lines/appearances work and none of that is covered in the manual. Thanks for the pointer though. I'd run into on one visit, but couldn't find it again later when I went back into the site. Be seeing you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /dev/zap not created
there might be an easier way, but i changed the asterisk source code before i compiled to reference the zap devices under /dev in their new place. i think everything you need to change is under the apps dir in the source tree On Wed, 2 Mar 2005 20:02:33 +0500, Rizwan Chaudhry [EMAIL PROTECTED] wrote: I installed asterisk on Fedora Core 2 kernel 2.6.5. I followed the standard procedure. zaptel-libpri-asterisk. The thing is that I constantly get the error message: line 4: Unable to open master device '/dev/zap/ctl' where the file zaptel.conf contains only 4 files: fxoks=1 fxsks=4 defaultzone=us loadzone=us I cant run asterisk and get a load of error messages. When I tried to check the directory /dev/zap, it wasnt there. It isnt created during installation. Can someone help me out. Apart from that, should I get a dialtone in the fxs module when the tdm400p is connected in the computer? because I have no way of checking whether my device is correctly connected. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!
Same problems as you... Eyebeam is not really fine in video... We have find some nasty bugs in it (PC freeze, codecs issues...) and no feedback from Xten after sending back reports (tcpdump and long descriptions). I think that EyeBeam works fine with... eyebeam. The software seems to be beta because of each version of Eyebeam I've download has differents bugs. Try with our hard-videophone ( ;-) ), Asterisk video features works. Perhaps a small problem in Intra Frame request (I've posted it in feature request without success). We will work on it ASAP. Nicolas http://www.call.fr -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Shadow Roldan Envoye : mardi 1 mars 2005 20:40 A : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] Broadvoice + Videosupport=yes - Fails! Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ We're sorry your call cannot be completed at this time So... I've commented it out and tried adding videosupport=yes to specific extensions, now video doesn't work as eyebeam reports remote user does not support video but broadvoice works. Bizarre I'm running CVS v1-0-02/15/05 Any ideas? _ Shadow Roldan IT Manager Zero G Software, Inc. tel: +1.415.512.7771 x 306 fax: +1.415.723.7244 mailto:[EMAIL PROTECTED] www.ZeroG.com The leading provider of multiplatform software deployment solutions. _ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple lines
Thats a typical situation for me on a Cisco 7940 - I'm sure its the same as any phone. When a second call comes in I can put the first on hold (and they hear MOH) and if I want I can blind-transfer both parties into a meetme room and then go join them there if I want. I'm not sure how the other phone brands handle multiple incoming calls but they must do. Derek David Masure wrote: Hi, Question... Is there a way to receive two phone calls on the same phone, or, for example to receive a phone call, put the call in stand-by and then make another call and finally, why not put them all together in conference... Thanks David Masure ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why should I answer a Newbie question, therethick!
On Wed, 2005-03-02 at 11:01 -0500, Race Vanderdecken wrote: Why is it so difficult to just ignore any question with Newbie in it? Because if nobody reads their questions they won't get any answers, and until you read the question you don't know if it is an idiot question. Everyone has to start somewhere. At least the newbie found the list. Questions like I have ISDN lines do I need an interface card? are known as closed questions, they _only_ have 2 possible answers yes or no. Newbiness is not an excuse for asking inane questions. The worse you can do is kick sand in their face. I can think of many worse things, in the worst case you could give them a lesson in comparatives and superlatives. This list is owned by no one. Absolutely agree, that neither you nor I or any one else who has responded with the same views own the list. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to handle ROSE operation 34
On Mon, Feb 28, 2005 at 04:16:55PM +0100, Martin Knipper wrote: Hi, i am getting the follwing messages with asterisk 1.0.5 [...] Feb 28 16:13:05 VERBOSE[8899]: !! Unable to handle ROSE operation 34 [...] Can anybody gibe me a hint what is is about ? It shouldn't cause any problems. It's just a message to say that your telco is sending a facility IE that has some data inside of it that aren't supported in libpri. Matthew Fredrickson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MozPhone
Where did you get it? I was looking on the internet and couldn't find any link to install this Mozilla extension. Is it also possible to install it on Firefox? Thanks, Roman Zhovtulya -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of administrator tootai Sent: Mittwoch, 2. März 2005 16:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MozPhone Glenn A. Thompson a écrit : Hi, Is anyone using mozPhone? If so any feedback you can provide? Yes. For what I'm doing with it work. Could be improved. -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why should I answer a Newbie question, therethick!
On March 2, 2005 11:27 am, Steven Critchfield wrote: As a person who spent 9 hours in traffic last winter just to drive 15 miles due to idiots who should have just stayed home, I think your analogy breaks down. At some point, you either need to learn to drive or you pay someone else to transport you or your stuff. Same applies to computer work, either you can do the work yourself or you pay someone else to do it. Even your snow driver analogy works here, you either get yourself out of the ditch or you pay someone to do it for you. The payment is not always monetary. Sometimes the payment is just a showing of sufficient effort. Back to your snow driver analogy, if the driver in the ditch is just waiting in the car for you to come over and push them out without even attempting anything on their own, you would be less inclined to bother. You would be even less inclined to continue exerting your own effort if the driver was not cooperating or wasn't even interested in getting out to help push. /me cheers I could not have said it better myself. This needs to go in the FRONT PAGE of the Wiki and the archive link needs to be put in the topic of #asterisk. Hell Olle's weekly newbie reminder email needs this put in it, too. AMEN, brother, AMEN!!! -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Administration manual for Sipura-841?
If you contact Sipura and prove to them you are a service provider, they'll give you access to an area that contains the manual and profile compiler for the 841. I dropped them a note and they gave me all the details on how to prove I'm a service provider, but I'm really just an end user and don't feel like spending the effort trying to pretend I'm in the business of selling their stuff. It seems rather silly to me to prevent technical documentation for a technical product from getting to the people that buy it. sigh Be seeing you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!
Shadow Roldan wrote: Sip.conf attached and includes relevant configs in general + broadvoice + 1 extension. This is with video enabled and the config in which broadvoice fails. Again, changing to videosupport=no in general section and everything works fine. Ugh... I was way off-base. Currently 'videosupport' is global, and cannot be changed on a peer-by-peer basis. I'll post a patch to Mantis to correct that. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-biz] [Asterisk-Users] IAX2 web client that workswithg723 / g729. We got One
If anyone is interested in producing a VS.NET project file that can build an OCX component from the IAXClient source code I will pay for it (within reason). We can then put the project file into the iaxclient cvs to everyone can enjoy it. Geoff -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Tuesday, March 01, 2005 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-biz] [Asterisk-Users] IAX2 web client that workswithg723 / g729. We got One Andres sounds as if this is Andres's own development. He mentioned IAX, not IAX2. My guess is that he might have used one of the IAX GPL Libraries and source trees, based on iaxClient and not libiax2. It is possible that Andres is not aware of the GPL terms that he has to adhere to, if he wants to commercialize this product. The Source code for IAX Phone is available from Steven Sokol's Web site and from a few other places. Some links are below. http://www.sokol-associates.com/Body.asp?IncPage=IaxDownload.htm http://www.angelfire.com/falcon/babarnazmi/iaxclient/iaxclientocx.htm This is where the IAX Full source can be downloaded from: Asterisk ActiveX Component by Omar Carvajal [EMAIL PROTECTED] Copyright (C) 2001-2002, Omar Carvajal COMPILATION These instructions are made for the Microsoft Visual C++ 6 compiler. Add the following directories in the include path: gsm\inc gsm\src libiax\src Make sure the miniphone.h file is in the directory, this is available from the CVS version After adding those directories to the path, go ahead and compile the ActiveX from the Build-Build virt1800.ocx menu option. I have also included a MS Visual Basic project to test the ActiveX, available in the vbtest directory. If you have any questions, comments or anything of the sort send me a message at [EMAIL PROTECTED] or at miguelc55 on AOL instant messenger. The Asterisk ActiveX is distributed under GNU General Public License. http://lists.digium.com/pipermail/asterisk-users/2002-August/003756.html Seshu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!
Kevin P. Fleming wrote: Ugh... I was way off-base. Currently 'videosupport' is global, and cannot be changed on a peer-by-peer basis. I'll post a patch to Mantis to correct that. (replying to myself G) This patch will take some time to prepare, as the code in chan_sip that handles this stuff seems to be implemented in a very poor way, and the proper fix will be more invasive than just configuration option parsing. For now, treat 'videosupport' as global, and if you have a SIP peer that rejects INVITEs when they contain potential video streams, you are out of luck. Sorry :-( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: wctdm and two tdm cards
I have a server I'm working on here with two tdm cards in it. 4 FXS and 4FX0. Both cards work fine on their own. The problem lies with using both in the system at once. I have verified the IRQ's are fine. I have tried switching the slots the cards reside in, no luck though. I am using ACPI but not APM. I am using gentoo latest, with vanilla 2.6(.10) kernel and udev. CVS as of CVS-HEAD-03/02/05-03:42:41. The problem is as follows: If I power up the system from system off, the cards both get detected If I reboot the system with reset button, ctrl alt del, or 'reboot' the TDM04P does not get detected. If I then reboot, then hit the power button, and let it turn off, then turn it back on again and boot, it detects both cards fine. I have tried searchign the list archives, but I have not had much luck. One person on IRC mentioned he's seen this before, but didn't have any solutions. Does anyone here know what might be the problem? or have a fix/work around? I know I shouldnt be rebooting servers, but I have to make sure it works upon reboot as it is going to be installed in a power-outtage happy part of the world :) I'm not having any problems like that with RHv9 (2.4 kernel), so I'd have to guess the issue is 'timing' related in whatever script that loads your tdm-zaptel drivers. As I recall (as a non-v2.6 user), there was an issue with timing and someone added a sleep/wait statement in the startup script to bypass the problem. Might consider finding your startup and add some additional time to that sleep/wait. Another approach to isolating the problem is to load the drivers by hand paying close attention to error messages, delays, etc. If your not sure how to do that, read your startup script and simply do those steps manually. Someone mentioned unplugging power and/or removing the card. That approach is totally BS. The same startup process is run regardless of whether one is rebooting or starting from power-on. There is nothing on the tdm card that stores values (no flash, no battery backup mem, etc). If the startup script operates one time from any startup mode, it is setting the tdm registers, etc, correctly. I'm away from the office this week, but I recall there was a readme shipped with the zaptel source that discusses kernel 2.6 timing issues. Might look for that in your src directory. The approach of non-rebooting but power off and then power on (cold reboot), is NOT BS. For me had been the only way to reboot the server with the TDM cards. If you make a hot reboot (al least with my cards) the modprobe will have fatal errors and won't load the cards. I think it's MoBo related, maybe some kind of IRQ assignment not released. = Miguel Ruiz Velasco Version: OpenKeyServer v1.2 Comment: Extracted from belgium.keyserver.net Signature: 0x59831109 __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MozPhone
On Wed, 2005-03-02 at 17:36 +0100, Roman Zhovtulya wrote: Where did you get it? I was looking on the internet and couldn't find any link to install this Mozilla extension. Dare I say google for Mozphone will give the link to French Polynesia from where it can be downloaded Is it also possible to install it on Firefox? Yes -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why should I answer a Newbie question, therethick!
If some one would like to show me the site that explains how to setup a mailing list then I will create a Newbie list for asterisk and voip questions and answers. I am only asking for someone to show me the site and maybe a few pointers on how to start it up. Only because I don't have the time or experience to do it quickely enough to get the newbies off the list. And I am a bit slow with apache and web type sutff, as you can tell by my website codetyrant.com. I will personally pay for the hosting of the list. It is not that I am tired or will ever grow tired of passing out fish and giving fishing lessons it is just I don't have the good fortune to be adept at web interfaces. Also, suggestions for the domain name would be welcomed. Race The Tyrant Vanderdecken In the Land of the Blind, the One-eyed man is Elvis..., copyright RPV 1997. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, March 02, 2005 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Why should I answer a Newbie question,therethick! On Wed, 2005-03-02 at 11:01 -0500, Race Vanderdecken wrote: This list is owned by no one. Actually it is owned by Digium. It has many contributers though. When someone drives off the road into a ditch in a snow storm they last thing they need is someone telling them they should have invested in snow chains and defensive driving lessons before leaving the house. Newbies need help getting out of the ditch so traffic can continue to flow and the rubber neckers can be abated. If you are not willing to pull off to the side of the road and help the fool by pushing their car out of the ditch you have no right to give him the finger as you drive past. As a person who spent 9 hours in traffic last winter just to drive 15 miles due to idiots who should have just stayed home, I think your analogy breaks down. At some point, you either need to learn to drive or you pay someone else to transport you or your stuff. Same applies to computer work, either you can do the work yourself or you pay someone else to do it. Even your snow driver analogy works here, you either get yourself out of the ditch or you pay someone to do it for you. The payment is not always monetary. Sometimes the payment is just a showing of sufficient effort. Back to your snow driver analogy, if the driver in the ditch is just waiting in the car for you to come over and push them out without even attempting anything on their own, you would be less inclined to bother. You would be even less inclined to continue exerting your own effort if the driver was not cooperating or wasn't even interested in getting out to help push. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More NAT questions
In you asterisk sip.conf: [general] externip=xxx.xxx.xxx.xxx ;ip address of your nat firewall (public ip) localnet=192.168.0.0/24; the local subnet where the asterisk box is If you don't externip, externip will never be used, because asterisk won't know WHEN to use it. Also, define canreinvite=no in your sip phones sections, as was suggested above. Julian J. M. On Wed, 2 Mar 2005 23:26:56 +1100, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Still trying to get NAT working. I have following setup: PHONE 1 -- * BOX | NAT/Firewall | | NAT/Firewall | | PHONE 2 Firewall next to phone 2 has all ports open. Firewall next to Asterisk has open ports 5060 and 1:2. All of those are forwarded to Asterisk box. Both phones succesfully register with Asterisk. (I had to add NAT=yes to configuration of PHONE 2 in sip.conf to get this far). Now, problems: I can place a call from PHONE2 to PHONE1, but sound path is not established. Calls from PHONE1 to PHONE2 can not be placed at all. (I assume that this is because port 5060 is not forwarded to the phone at NAT/Firewall, but more on it later). Looking at SIP debug info, Asterisk tries to use local address of PHONE2 instead of its public IP. As a result, no info can be sent to it. I have tried to install SIPROXD on the NAT/Firewall close to Asterisk box, but this did not help. Now, we have tried to use one of the commercial VoIP service at PHONE2 location. We had to use their phone and it worked just fine without any alterations to NAT/Firewall device. I am pretty sure that they use SIP, so they did resolve the problem somehow. Sorry, there is no technical info available on this service. Did anyone succeeded in doing this setup? I know, IAX is a better way, but I can not setup many Asterisk boxes. Basically, I am doing it for a friend. He is working for a small medical company. They have number of offices that are not open every day and offices are too small to put Asterisk box in each one. There will be 1-3 IP phones in each office, except central one. Central one will need Asterisk, the rest should be on their own. Any help is greatly appreciated. Thanks, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MozPhone
Roman Zhovtulya a écrit : Where did you get it? I was looking on the internet and couldn't find any link to install this Mozilla extension. Have a look at: http://taina.sysnux.pf:8080/cps/sections/telephonie/copy_of_mozphone/view Just clic on the link in the install section and install should begin, except it might be blocked by Mozilla (or Firefox) security settings. Is it also possible to install it on Firefox? Yes, it definitely works with Firefox. Thanks, Jean-Denis Girard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial application invoked again and again
hi all i am using CVS with Realtime mysql on backend. Dial application is invoked again and again what is the reason. I have tested it by printing some message to debug. this application is invoked again and again here is debug you can see lot of messages from app_dial.c at the end. Any one tell me what is the reason. Is this a bug or what Kamran Ahmad -- *CLI sip debug SIP Debugging Enabled *CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.117;branch=z9hG4bK2038176231 From:sip:[EMAIL PROTECTED]; To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE Contact: sip:[EMAIL PROTECTED] Max-Forwards: 5 User-Agent:SKYPHONE/1.03 Subject: hello Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER,SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length:180 v=0 o=sibtay 2890844 842807 IN IP4 192.168.0.117 s=SDP Seminar c=IN IP4 192.168.0.117 t=0 0 m=audio 13044 RTP/AVP 0 101 a=rtpmap:101 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:96 0-11,16 14 headers, 10 lines Using latest request as basis request Sending to 192.168.0.117 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.117:13044 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Found user '3000' Looking for 2000 in default list_route: hop: sip:[EMAIL PROTECTED] Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.117;branch=z9hG4bK2038176231 From: sip:[EMAIL PROTECTED]; To: sip:[EMAIL PROTECTED];tag=as7a83cce0 Call-ID: [EMAIL PROTECTED] CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.117:5060 Mar 3 10:44:01 WARNING[6311]: app_dial.c:618 dial_exec_full: hello i am from app_dial We're at 192.168.0.203 port 15344 Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x2 (gsm) Answering with capability 0x8 (alaw) 12 headers, 10 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.203:5060;branch=z9hG4bK56922e05 From: 3000 sip:[EMAIL PROTECTED];tag=as35d782e5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 03 Mar 2005 05:44:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 207 v=0 o=root 6311 6311 IN IP4 192.168.0.203 s=session c=IN IP4 192.168.0.203 t=0 0 m=audio 15344 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - (no NAT) to 192.168.0.117:5060 Sip read: SIP/2.0 486 Busy Here From:sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED] Contact:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: SKYPHONE/1.03 via: SIP/2.0/UDP 192.168.0.203:5060;branch=z9hG4bK56922e05 Content-Length: 0 9 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.203:5060;branch=z9hG4bK56922e05 From: 3000 sip:[EMAIL PROTECTED];tag=as35d782e5 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.0.117:5060 Destroying call '[EMAIL PROTECTED]' Mar 3 10:44:11 NOTICE[6311]: rtp.c:452 ast_rtp_read: RTP: Received packet with bad UDP checksum Mar 3 10:44:11 WARNING[6311]: app_dial.c:618 dial_exec_full: hello i am from app_dial Mar 3 10:44:11 WARNING[6311]: chan_sip.c:1345 create_addr: No such host: t Destroying call '[EMAIL PROTECTED]' Mar 3 10:44:11 NOTICE[6311]: app_dial.c:918 dial_exec_full: Unable to
Re: [Asterisk-biz] [Asterisk-Users] IAX2 web client that workswithg723 / g729. We got One
That's exactly the point. I think it's 90% likely that Andres' project uses iaxclient, and 98% likely that it uses libiax2. If this is the case, he needs to comply with the licensing terms of these libraries. Specifically, he would need to make the source code to his version of these libraries publicly available. Most people who work with iaxclient know that I'm not personally interested in much of anything windows only, and I don't use any Microsoft tools to develop things, but I do want to see people collaborate, and I specifically do _not_ want to see people taking without at least making their work available, as required by the license. I'd probably prefer even more if some of the people working on different GUIs would team up so that instead of having X different GUIs, there were X/3 GUIs that were 3 times as good, but I suspect that consolidation will happen in time.. As far as this particular case, I could be wrong, and Andres could have written everything from scratch, but, the sentiment would apply either way. Geoff Nordli wrote: If anyone is interested in producing a VS.NET project file that can build an OCX component from the IAXClient source code I will pay for it (within reason). We can then put the project file into the iaxclient cvs to everyone can enjoy it. Geoff -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Tuesday, March 01, 2005 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-biz] [Asterisk-Users] IAX2 web client that workswithg723 / g729. We got One Andres sounds as if this is Andres's own development. He mentioned IAX, not IAX2. My guess is that he might have used one of the IAX GPL Libraries and source trees, based on iaxClient and not libiax2. It is possible that Andres is not aware of the GPL terms that he has to adhere to, if he wants to commercialize this product. The Source code for IAX Phone is available from Steven Sokol's Web site and from a few other places. Some links are below. http://www.sokol-associates.com/Body.asp?IncPage=IaxDownload.htm http://www.angelfire.com/falcon/babarnazmi/iaxclient/iaxclientocx.htm This is where the IAX Full source can be downloaded from: Asterisk ActiveX Component by Omar Carvajal [EMAIL PROTECTED] Copyright (C) 2001-2002, Omar Carvajal COMPILATION These instructions are made for the Microsoft Visual C++ 6 compiler. Add the following directories in the include path: gsm\inc gsm\src libiax\srcMake sure the miniphone.h file is in the directory, this is available from the CVS version After adding those directories to the path, go ahead and compile the ActiveX from the Build-Build virt1800.ocx menu option. I have also included a MS Visual Basic project to test the ActiveX, available in the vbtest directory. If you have any questions, comments or anything of the sort send me a message at [EMAIL PROTECTED] or at miguelc55 on AOL instant messenger. The Asterisk ActiveX is distributed under GNU General Public License. http://lists.digium.com/pipermail/asterisk-users/2002-August/003756.html Seshu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax with spandsp + zaphfc
Hi all, I'm trying to enable faxes on my asterisk box using spandsp version pre10. The outgoing and incomiong calls work great, but i have problem with faxes. The fax detection works, the RxFax application is called, but i only receive the beginning of the faxes (variable between 3 to 5 first inches of the fax). The output .tif file generated is then cut off. My setup is : asterisk 1.0.6+zaphfc driver from bristuff RC7j. Bewan BRI PCI card. zapata echo cancellation is disabled. Where is some log snippets : Mar 2 16:05:33 VERBOSE[12844]: -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/fax/asterisk-12844-1109775932.0.tif) in new st ack Mar 2 16:06:38 DEBUG[12844]: = = Mar 2 16:06:38 DEBUG[12844]: Pages transferred: 1 Mar 2 16:06:38 DEBUG[12844]: Image size: 1728 x 82 Mar 2 16:06:38 DEBUG[12844]: Image resolution7700 x 3850 Mar 2 16:06:38 DEBUG[12844]: Transfer Rate: 9600 Mar 2 16:06:38 DEBUG[12844]: Bad rows71 Mar 2 16:06:38 DEBUG[12844]: Longest bad row run 34 Mar 2 16:06:38 DEBUG[12844]: Compression type2 Mar 2 16:06:38 DEBUG[12844]: Image size (bytes) 0 Mar 2 16:06:38 DEBUG[12844]: = = Mar 2 16:06:41 DEBUG[12844]: = = Mar 2 16:06:41 DEBUG[12844]: Fax successfully received. Mar 2 16:06:41 DEBUG[12844]: Remote station id: 0142401377 Mar 2 16:06:41 DEBUG[12844]: Local station id: Mar 2 16:06:41 DEBUG[12844]: Pages transferred: 1 Mar 2 16:06:41 DEBUG[12844]: Image resolution: 7700 x 3850 Mar 2 16:06:41 DEBUG[12844]: Transfer Rate: 9600 Mar 2 16:06:41 DEBUG[12844]: = = Do anyone of you have any experience with spandsp on zaphfc compliant card ? Thanks a lot Thibault ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MozPhone
administrator tootai a écrit : Glenn A. Thompson a écrit : Hi, Is anyone using mozPhone? If so any feedback you can provide? Yes. For what I'm doing with it work. Could be improved. Thanks for your feedback. MozPhone could obviously be improved in many ways, what would be your suggestions? Thanks, Jean-Denis Girard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Looking for asterisk integrators in Dallas,TX
Sorry for posting this OT: If you are an asterisk integrator in the Dallas Area or are willing to travel for a Presentation please mail me to [EMAIL PROTECTED] Thank you, Victor Perez ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk HEAD and Mysql problems
Guys.
I just updated my asterisk with the current HEAD from the cvs and everything
compiled great, asterisk, the addons, etc.
But I just checked one MYSQL app Im using and I got an error, seems asterisk
cant connect to mysql anymore.
-- Executing MYSQL(SIP/casa1-926c, Connect connid localhost xxx
asterisk) in new stack
Mar 2 11:39:48 WARNING[3443]: pbx.c:1357 pbx_extension_helper: No
application 'MYSQL(Query resultid ${connid} SELECT\ id\ FROM\ demousers\
WHERE\ user=\'${user}\'\ and\ password=\'${password$' for extension
(intruder, 5, 6)
but MYSQL does show on show applications list and mysql cdr IS working.
Any hints?
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Re: [Asterisk-Users] MozPhone
Jean-Denis Girard wrote: Roman Zhovtulya a écrit : Where did you get it? I was looking on the internet and couldn't find any link to install this Mozilla extension. Have a look at: http://taina.sysnux.pf:8080/cps/sections/telephonie/copy_of_mozphone/view Just clic on the link in the install section and install should begin, except it might be blocked by Mozilla (or Firefox) security settings. FYI the above link does not work. The host is not found ( at least by our nameservers ) but the Google.com search found this. http://www.sysnux.pf/cps/sections/telephonie/copy_of_mozphone/switchLanguage/en The only issue is the download/install link doesn't work. I have sent a message to the webmaster. Jean-Denis -- do you have this honor? Rod -- --- [This E-mail scanned for viruses by Declude Virus] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MozPhone
DEAD link... cannot be located On Wed, 2005-03-02 at 07:18 -1000, Jean-Denis Girard wrote: Roman Zhovtulya a écrit : Where did you get it? I was looking on the internet and couldn't find any link to install this Mozilla extension. Have a look at: http://taina.sysnux.pf:8080/cps/sections/telephonie/copy_of_mozphone/view Just clic on the link in the install section and install should begin, except it might be blocked by Mozilla (or Firefox) security settings. Is it also possible to install it on Firefox? Yes, it definitely works with Firefox. Thanks, Jean-Denis Girard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- skamp [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP300 soft key configuration
I'm trying to reconfigure my IP300 softkeys.. Currently when on a call, I have to hit more and then transfer.. I'd like make transfer appear on the first screen. Right now there's hold on there.. and hold is kind of redundant, since the IP300 has a hard hold button. I tried doing it in the keys/ section of ipmid.cfg, but it doesn't seem to work.. anyone done this or something similar? I checked the admin guide but am confused.. I think I'm supposed to be using the key.x.y.function.prim thing, but it doesn't work for me.. Thanks in adavance.. --Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
