Re: [Asterisk-Users] diax softphone
Greg, my Linux iptables firewall, on a private network. Both boxes cann register iax2 to asterisk, and dial, but as soon as asterisk tries to do the native a private network -- as in a NATed network? Maybe canreinvite=no or nat=yes will do the magic you need. I think he is using the IAX2 protocol, and I can't find anything in the sources that indicate canreinvite or NAT in the sources of chan_iax.c. BTW: I have the same problem. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] diax softphone
Hi, - Original Message - From: Peer Oliver schmidt [EMAIL PROTECTED] Greg, my Linux iptables firewall, on a private network. Both boxes cann register iax2 to asterisk, and dial, but as soon as asterisk tries to do the native a private network -- as in a NATed network? Maybe canreinvite=no or nat=yes will do the magic you need. I think he is using the IAX2 protocol, and I can't find anything in the sources that indicate canreinvite or NAT in the sources of chan_iax.c. BTW: I have the same problem. I have 2 DIAX phones behind two different NAT firewalls and the * box on one of the phones network. It works for me. One of the NATs is a Wndows RRAS and the other one is a hardware broadband router from Netgear. I have just opened the 4569 UDP port on the firewall in both direction (for input it is forwarded to the * box). BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] diax softphone
Hi Dan, iax2 to asterisk, and dial, but as soon as asterisk tries to do the native BTW: I have the same problem. I have 2 DIAX phones behind two different NAT firewalls and the * box on one of the phones network. It works for me. Cool. I am sure it has nothing to do with DIAX, but might be the configuration on the * side. Using IAX(1) works fine, btw. I have 4569 opened and forwarded/NATed to my *. I am on the same network as the * server, a friend is remote. After about a minute you loose the connection. rgds pos ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] diax softphone
Hi, I have 2 DIAX phones behind two different NAT firewalls and the * box on one of the phones network. It works for me. Cool. I am sure it has nothing to do with DIAX, but might be the configuration on the * side. Using IAX(1) works fine, btw. This is very interesting. It seems they still are a lot of bugs to be solved in IAX2 ...:-( I have 4569 opened and forwarded/NATed to my *. I am on the same network as the * server, a friend is remote. After about a minute you loose the connection. This is another problem and it happens for me too (known bug, which seems to be related with the re-registration which occurs at 60s). BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] billing information from telecom
Hi everyone, I have TE410P connected via E1 to telecom and another E1 to my internal Ericsson PBX. All calls from PBX to telecom pass through Asterisk. Our telecom is providing us with billing information during a call, and I would like to transfer this information to PBX and be able to show it on user displays. If I connect my PBX directly to telecom trunk it is working ok, but now, with Asterisk in between I am not getting it. Can anyone tell what should be done regarding this. thanks Tomica
RE: [Asterisk-Users] 8 lines - best approach
The problem is when replacing a Nortel system. The existing phones become useless, so we're looking at either using totally IP based phones or using a channel bank with different office phones. The only problem is finding an IP phone that is decent for business, supports multiple lines (at least 2) and is reasonably priced. This is more difficult than I expected. The voip-info.org site has excellent information, but it seems to show the IP phone coverage is still in the early adopter stage. Please correct me if my statement is wrong, I'd love for it to be wrong. My thinking is that most of us are looking for hardware based IP Phone's that work well with Asterisk and begin to go beyond the traditional business phone. So my first step is to find the most cost effective way to utilize our traditional lines and move on from there. Cheers, Darren -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Saturday, January 24, 2004 5:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 8 lines - best approach On Friday 23 January 2004 12:18, Paul Mahler wrote: On Friday, January 23, 2004 at 8:04 AM, Steven Critchfield wrote: On Fri, 2004-01-23 at 09:30, Darren Martz wrote: I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be roughly $2600 for 8 lines given that system - perhaps my estimate is way off Is there another way that is more cost effective? That number sounds about right. It is likely that it will be less, but budgeting that much for hardware is a good start. Do you have to continue to use the existing handsets? You should look at replacing the existing phones with SIP phones. He did say cost-effective. Last I checked, 24 SIP phones (unless they are Grandstreams) will cost far more than a channel bank. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Dialling Hook Flash on Zaptel
I had some similar problems with the X100P and our ATA-2. I also couldn't ever get the Nortel to recognize the DTMF, or get Asterisk to recognize DTMF coming through the Nortel. I wish I could say that I figured out a really cool way to make it work, but instead I moved on and interconnected via PRIs. I did a little more testing here, I've found that from my Cisco 7940 dialing out to my mobile, I can dial DTMF tones and hear them on the mobile. I'm not sure if the Norstar is doing this, as no matter how long I press the button down for I only get a short beep of the DTMF tone on the mobile. Perhaps this means the Norstar can only pass along the tones but not actually interpret them, or maybe the DTMF tone length is too short for the Norstar. Either way, I've changed the station filter for this particular extension to allow a greater range of numbers to be dialled and will control it with the dial plans in Asterisk. I've also considered changing the interconnection method, unfortunately (although this may be a good thing) my system is only a baby CICS with a 4-port analog trunk module and a 4-port BRI module. To connect via the analog trunk would be really neat with a 4-port FXS digium card, but unfortunately this particular Nortel card is not a supervised card, so can't be setup in the Norstar for auto-answer (which was my main reason for installing the BRI card). Then in terms of connecting via BRI, I think it would probably be more effective in the long run to just replace the whole system with SIP handsets since there's only 7 extensions in use (although the cost of Cisco 7940's would quickly add up, but I wouldn't want to use anything less even though they may be cheaper, as these are fantastic phones and really worth it IMHO). Also apart from the handset replacement cost, I think it'll be somewhat hard to beat the near bullet-proof performance the current CICS system has given us. The only outages its ever had was to install the BRI card and the odd power outage that was long enough to fully drain the UPS batteries. Cheers, Chris Lee ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] GS and NAT
hi. I've gs working under NAT, simply put nat=yes into sip.conf section if *, then enable nat into the gs, without any stun server. Matteo. Il mar, 2004-02-03 alle 21:17, Tomas Prybil ha scritto: Hi all. Is it at all possible to have a GS B101 NATed with firmware 1.0.4.40? I've tried both STUN and not STUN. The odds seems best with stun because the phone registers with right ip adress. When the connection is made * sends rtp packets to the right destination AND port, but the phone doesn't accept the packets. Should I burn my D-LINK 604 or upgrade the GS? /t ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P and PSTN line Callwaiting
Hi, There is any way to use the PSTN line callwaiting functionality (including callwaiting callerID) with an X00P card? When a second incoming call, on an internal ATA I hear the callwaiting tone, but I don't know how to switch to the other caller through ATA-*-X100P. More, the callerid is not displayed during callwaiting (just during a standard call). There is any direct way to switch the calls from the phone connected to an ATA device? There is any modification to be made in the callerid.c file for the callwaiting callerid functionality? (I have to make one for the normal call in order to support my provider type of callerid signal, but does not work for callwaiting callerid too). Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP debug logs
Debuuging SIP to a file: asterisk -c | tee /tmp/sipdebug.log then turn on 'sip debug' at the CLI /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Do you Linux softphone..
An article I came across this morning.. http://www.itnews.com.au/storycontent.asp?ID=12Art_ID=18128 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] IAX2 (was: diax softphone )
Hi, From: Peer Oliver schmidt [EMAIL PROTECTED] I have 4569 opened and forwarded/NATed to my *. I am on the same network as the * server, a friend is remote. After about a minute you loose the connection. This is another problem and it happens for me too (known bug, which seems to be related with the re-registration which occurs at 60s). Maybe those two problems are related to each other, i.e. IAX2 tries to bridge the call ... (I have no idea what I am talking about) I think that the call is allways passed through the * server if IAX(2) is used. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Code Hosting...
lo, Is there a single central location for code and applications other than CVS? I'm talking about code that can't/wont be included in CVS for various reasons? Does the wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has done some updates too) but apparently this will never make it into CVS (since the engine is not GPL)... Seems to make sense to have a central location for this type of 'outlaw' code... The bug tracker is useless for this sort of thing but there seem to be a number of bits of code like this in there moo Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Code Hosting...
Andy Powell wrote: lo, Is there a single central location for code and applications other than CVS? I'm talking about code that can't/wont be included in CVS for various reasons? Does the wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has done some updates too) but apparently this will never make it into CVS (since the engine is not GPL)... Seems to make sense to have a central location for this type of 'outlaw' code... The bug tracker is useless for this sort of thing but there seem to be a number of bits of code like this in there moo Andy I don't know of any repository, if you have access to a webserver why not start one? :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Code Hosting...
Andy, I would be interested in your Cepstral engine code. Regards, Robert Friedrichshafen, Germany Andy Powell said: lo, Is there a single central location for code and applications other than CVS? I'm talking about code that can't/wont be included in CVS for various reasons? Does the wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has done some updates too) but apparently this will never make it into CVS (since the engine is not GPL)... Seems to make sense to have a central location for this type of 'outlaw' code... The bug tracker is useless for this sort of thing but there seem to be a number of bits of code like this in there moo Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Port bind
I have two cards in one of the servers. If I bind SIP port to public IP, it all works fine. If I do not bind to specific IP (ie. Bind = 0.0.0.0), I get segmentation fault while starting *. Can SIP (and other protocols), bind to more then one IP address? If yes, what is syntax? SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Port bind
G'day, On Wed, 4 Feb 2004, Senad Jordanovic wrote: I have two cards in one of the servers. If I bind SIP port to public IP, it all works fine. If I do not bind to specific IP (ie. Bind = 0.0.0.0), I get segmentation fault while starting *. I have this configuration also, with two network cards and sip.conf specifying bindaddr = 0.0.0.0. No problems, so it definitely works. I think you'll need to provide further debug information (logs, core analysis, etc) so that the gurus can try and help you. Hoo-roo, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] IAX2 (was: diax softphone )
On Wed, 4 Feb 2004 11:13:52 +0200, Dan wrote Hi, From: Peer Oliver schmidt [EMAIL PROTECTED] I have 4569 opened and forwarded/NATed to my *. I am on the same network as the * server, a friend is remote. After about a minute you loose the connection. This is another problem and it happens for me too (known bug, which seems to be related with the re-registration which occurs at 60s). Maybe those two problems are related to each other, i.e. IAX2 tries to bridge the call ... (I have no idea what I am talking about) I think that the call is allways passed through the * server if IAX(2) is used. BR, Dan Actually, IAX(2) calls are bridged whenever they can. The server sends TXREQ to both legs of the call and the clients try to connect to each other. What came to my mind is that it might break in NAT environment... Let's see (Note: not tested in the wild, just speculations): - UA1 10.0.0.10:4569 - inside NAT - UA2 1.1.1.1:4569 - outside NAT - server 10.0.0.1:4569/2.2.2.2:4569 - accessible from both UAs UA1 and UA2 register on the server, which fills their apparent_address info the way it can see them (UA1 has private IP, UA2 is public). When a call is established and UA1 and UA2 send TXCNTs to each other, they get the peer address from the * server, so - UA1 tries to connect to 1.1.1.1:4569 which should work (NATted by the router) - UA2 tries to connect to 10.0.0.10:4569 - no go. - UA1's packet reaches UA2 (say, NATted to 2.2.2.2:5), so UA2 sends TXACC, but (correct me if I'm wrong here) the peer address is not updated. Thus, the connection breaks as UA1 never receives the TXACC that UA2 sent. If libiax2 (and chan_iax2 - though I haven't actually looked into it too much) set the peer's address upon receipt of TXCNT, it might work (UA2 would talk to 2.2.2.2:5 which the router would NAT back to 10.0.0.10:4569). If that's not the problem, sorry for confusing you. :) A packet dump should reveal all. What do you think? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] IAX2 (was: diax softphone )
Hi, - Original Message - From: Grzegorz Nosek [EMAIL PROTECTED] ... A packet dump should reveal all. What do you think? Good idea. Use Debug feature in DIAX and send both phones and asterisk logs to check them offline (you'll have timestamps on each phone log so you can rebuild the whole conversation. As I told before, it works in my environment, so I have no way to reproduce this behaviour here. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie: Chan_capi, early b3 in Italy
When I make a call and the other party is busy I do not hear anything
but a free ringing phone.
Also, if I call a call center with a voice menu the phone keep ringing
without any sign of life.
I tried early b3 with these ones but nothing change much:
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+CAPI+readmediff=2
My extensions are
[outgoing]
exten = 0,1,Goto(outgoing-isdn,s,1)
[outgoing-isdn]
exten = s,1,NoOp()
exten = _X.,1,Dial(CAPI/0255:b${EXTEN}|30)
Any idea?
Is Asterisk compatible with Italian signal out of the box?
Thank you all in advance
mr
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RE: [Asterisk-Users] Minor Registration Problem With Polycom Soun dpoint IP 500
What firmware and sip versions are you using? I have several Polycom phones on my system right now and I've never had any registration problems with them. Instead of leaving the host as dynamic try declaring an IP address(that's the only difference I see between your sip.conf and mine). If you are still having problems I've like to see your polycom .cfg files for one of these phones, you might be missing a setting in one of them. MATT--- -Original Message- From: David Liu [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 1:06 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Minor Registration Problem With Polycom Soundpoint IP 500 We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk environment. So far it has been good. Call Hold, Transfer, DMTF etc. However, I do notice every now and then the Polycom fails to register with Asterisk. Asterisk console outputs the following: Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response: Unable to determine sequence number from '' Feb 3 13:02:34 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to authenticate user DavidLiu sip:[EMAIL PROTECTED];tag=9F67E426-59D92ED7 Feb 3 13:02:36 NOTICE[278546]: chan_sip.c:5125 handle_request: Failed to authenticate user DavidLiu sip:[EMAIL PROTECTED];tag=BFDEF35B-1CBC4F2C in sip.conf: canreinvite=yes host=dynamic canreinvite=yes dtmfmode=rfc2833 context=sip port=5060 Usually say after the phone failed to register with Asterisk, I can attempt to place a call. It will fail of course. But then I can try calling again and usually the call will go through and it will successfully re-register itself without needing a restart. What can this be? Surely Polycom is re-registering every 3600 before Asterisk times it out. But Asterisk is just refusing it. By the way, anyone know whether Asterisk is geared towards RFC3261 or RFC2543? I know Asterisk is not a fully SIP Proxy but lets say if a SIP PSTNGW or a SIP phone is designed under the spec 2543 as suppose to 3261, will it work better or the same with Asterisk? David ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] VOIP Deployment Concerns
On Wed, Feb 04, 2004 at 12:38:37AM -0500, William Suffill said: I will be moving in a few months and I'm concerned as to what kind of bandwidth I would need to work effectively. The reason I posed the question here is simple most of my work is remote SSH to various BSD/Linux machines but a majority of my business calls from the office and clients will be routed through Asterisk. Currently I use a SIP phone snip Where I'll be moving will more than likely be installing broadband from Adelphia Cable 256up 3mbit down unless I have a valid reason to require the 512up/4mbit down prem. package at approximately 80 a month. I don't have any personal experience with them since I do live in NJ and will be relocating to FL. Any advise would greatly appreciated. snip Good luck with Adelphia. My experience is that the service is VERY flakey. Sure, you can occasionally get the 3Mb down, rarely get the 256Kb up, but the norm seems to have LOTS of service outages that last 5 seconds to a minute. I also found that as soon as school let out, bandwidth and latency SUCKED. It also frequently went offline for hours at a time (3 days in one case.) My normal upload was 5Kb, and download was 100Kb. IMHO, cable is not a good medium for VOIP (at least adelphia's service... I've heard other cable companies are better.) When called about it, Adelphia claims that their service is designed for web browsing and not interactive applications. WTF??? It would be interesting to hear others experiences. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] CALEA?
What are my support options for CALEA with Asterisk? Don't think you'll find any support without some major AGI/Manager programming, and then not likely to interface with the calea requesters. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 quick dial
Out of interest, does anyone know if it's possible to get the 7960 to start accepting a number while on-hook, without having to press NewCall, the line button, or speaker button? This is just something I was used to with the Norstar extensions, I could immediately start dialing the numbers for an internal extension and it'd work. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jose Inzunza/YM/RWDOE Sent: Wednesday, 4 February 2004 2:21 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 quick dial Is there a way to make the Cisco 7960 SIP phone dial out automatically without having to press the dial button, once the numbers that you have entered match a specific pattern? This feature is present when the phone is working with a Cisco CallManager. For example, if all of my internal extensions begin with a '5' and are four digits long, if I dialed '5123' on the phone, the call would initiate once I pressed the '3'. Any help would be appreciated. Jose ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Mediatrix sip fxo gateway workaround?
Rich, If the Mediatrix uses the Caller-ID field to select which channel to use, then you really have no choice but to do that. As you pointed out, the Caller-ID info is not (and cannot) be passed to the PSTN line. Rich Adamson wrote: Possible Mediatrix 1204 fxo sip gateway workaround Need some feedback from experienced * users relative to this workaround please please please. Problem: The mediatrix 4-port fxo gateway does not provide any mechanism for * to select which port an outbound pstn call will use. (See lots of previous posts over the past four days for more detail if needed.) Our reseller has been working with Mediatrix to find a way for * to send pstn calls to a specific port number on their 1204 4-port fxo sip gateway. The proposed work around (below) sets a unique-but-well-known CallerID prior to sending the call to the 1204, and the 1204 filters on the CallerID sending the outbound call to the designated port. (The 1204 _does_ have such filtering/routing capability.) Since this unique callerid is _never_ forwarded to the US pstn providers, does anyone see any technical or management problem with using this approach both in the short and long term??? I'm thinking this is an acceptable workaround since it does not require micro-managing the dialplan, the 1204, etc. In my case, I'm not very concerned with scaling the solution since we could only hope business would increase to the point where four additional pstn analog lines were needed. ;) (FWIW, a 3,congestion statement can be added to the proposed statements.) Thoughts anyone? Rich Ok, you need to use the net2pstnsourcefilter to make this work. In this example you need to set port 1 to , 2 to , 3 to , 4 to . Then with the extension configuration below, and number starting with 9 will go to port 1 with the 9 removed from the string sent. Any number starting with 8 will be sent to port 2 with the 8 removed from the string sent and so on. It works like a charm on my 1204. [SIP] exten = _9.,1,SETCIDNUM() exten = _9.,2,Dial,SIP/[EMAIL PROTECTED] exten = _8.,1,SETCIDNUM() exten = _8.,1,Dial,SIP/[EMAIL PROTECTED] exten = _7.,1,SETCIDNUM() exten = _7.,2,Dial,SIP/[EMAIL PROTECTED] exten = _6.,1,SETCIDNUM() exten = _6.,2,Dial,SIP/[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 quick dial
Out of interest, does anyone know if it's possible to get the 7960 to start accepting a number while on-hook, without having to press NewCall, the line button, or speaker button? This is just something I was used to with the Norstar extensions, I could immediately start dialing the numbers for an internal extension and it'd work. I don't believe there is any way to do that. Would be handy though. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 bug in 6.1 evident in Asterisk
Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 : No one-way audio
One friend with Cisco 7960 with public IP address connect to my * box and I called me to my home phone through a X100P. He can hear me clearly and I cannot hear him. I thought the problem could be a NAT in the middle.. but there is no NAT. Any thoughts? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 quick dial
I have one word for you... LAZY! bkw On Wed, 4 Feb 2004, Christopher Lee wrote: Out of interest, does anyone know if it's possible to get the 7960 to start accepting a number while on-hook, without having to press NewCall, the line button, or speaker button? This is just something I was used to with the Norstar extensions, I could immediately start dialing the numbers for an internal extension and it'd work. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jose Inzunza/YM/RWDOE Sent: Wednesday, 4 February 2004 2:21 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 quick dial Is there a way to make the Cisco 7960 SIP phone dial out automatically without having to press the dial button, once the numbers that you have entered match a specific pattern? This feature is present when the phone is working with a Cisco CallManager. For example, if all of my internal extensions begin with a '5' and are four digits long, if I dialed '5123' on the phone, the call would initiate once I pressed the '3'. Any help would be appreciated. Jose ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Boards falling out...
I have a TDM40B, 4-port fxs card. Each port seems to have it's own little board on the fxs card. Each little board is not sodered in, but rather hangs (I have a vertical case for the server) on what I would call jumper pins (sorry, I'm not a profession geek, just a wannabe). One of my little boards, over time, slides off those jumper pins. I just noticed it this morning. I had to power down, seat it, and power up again. That's a pain. We did, though, have an earthquake this morning. That may have shaken things loose a bit. But, it wasn't much to speak of (long, but not strong). Has anyone else experienced this problem? What could I do to solve it (seat the little card a little more permanently)? Thanks ahead of time. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 quick dial
Then you've got to hand it to Nortel, they do know how to make a damn good phone extensions for lazy people like me :-) I actually believe this isn't the case with the Nortel Meridian systems, as I noticed when using one it wouldn't accept the numbers without first pressing that extensions DN key... perhaps it's just a Norstar CICS/MICS features. Anyway I've opened a TAC case with Cisco and will await their response, which I'm guessing already will be no, can't hurt to ask tho :-) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, 4 February 2004 11:58 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 quick dial I have one word for you... LAZY! bkw On Wed, 4 Feb 2004, Christopher Lee wrote: Out of interest, does anyone know if it's possible to get the 7960 to start accepting a number while on-hook, without having to press NewCall, the line button, or speaker button? This is just something I was used to with the Norstar extensions, I could immediately start dialing the numbers for an internal extension and it'd work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 quick dial
hehe ya I have to admit they are very featureful. :P Asterisk is still a baby i'm sure sip phones will get better with time. But you do have to admit that the cisco 7960's are damn good phones. bkw On Thu, 5 Feb 2004, Christopher Lee wrote: Then you've got to hand it to Nortel, they do know how to make a damn good phone extensions for lazy people like me :-) I actually believe this isn't the case with the Nortel Meridian systems, as I noticed when using one it wouldn't accept the numbers without first pressing that extensions DN key... perhaps it's just a Norstar CICS/MICS features. Anyway I've opened a TAC case with Cisco and will await their response, which I'm guessing already will be no, can't hurt to ask tho :-) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, 4 February 2004 11:58 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 quick dial I have one word for you... LAZY! bkw On Wed, 4 Feb 2004, Christopher Lee wrote: Out of interest, does anyone know if it's possible to get the 7960 to start accepting a number while on-hook, without having to press NewCall, the line button, or speaker button? This is just something I was used to with the Norstar extensions, I could immediately start dialing the numbers for an internal extension and it'd work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] talking clock
Hello I am looking for a AGI application that can say the current time with seconds, but i don't need the day/year. Has anyone got this already? Thanks in advance Deepak
Re: [Asterisk-Users] Code Hosting...
I agree. app_cepstral is a damn fine app and has been banished to the edges of the earth because the theta engine isn't open src. I even added a standalone build for app_cepstral... so you can download it.. make it and install it without much trouble. :( Andy maybe we can go thru and pickout the offending code and compile them on a site or something? I can setup a mirror for them. bkw On Wed, 4 Feb 2004, Andy Powell wrote: lo, Is there a single central location for code and applications other than CVS? I'm talking about code that can't/wont be included in CVS for various reasons? Does the wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has done some updates too) but apparently this will never make it into CVS (since the engine is not GPL)... Seems to make sense to have a central location for this type of 'outlaw' code... The bug tracker is useless for this sort of thing but there seem to be a number of bits of code like this in there moo Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] talking clock
SayUnixTime will do that just give it the format you want. SayUnixTime([unixtime][|[timezone][|format]]) unixtime: time, in seconds since Jan 1, 1970. May be negative. defaults to now. timezone: timezone, see /usr/share/zoneinfo for a list. defaults to machine default. format: a format the time is to be said in. See voicemail.conf. defaults to ABdY 'digits/at' IMp Returns 0 or -1 on hangup. bkw On Wed, 4 Feb 2004, Deepakumar JV wrote: Hello I am looking for a AGI application that can say the current time with seconds, but i don't need the day/year. Has anyone got this already? Thanks in advance Deepak ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 quick dial
Absolutely no argument from me on that front, hands down the Cisco 7940/7960 are a damn good IP phone, and compared to the existing Norstar handsets we have, a far better phone overall. The handsfree functionality on the Cisco's is truly awesome, the mic pickup and clarity is far better than the Norstar and people can barely tell the difference between talking to them on handsfree or picking up the handpiece. Definitely worth every dollar, although I wouldn't say no to them lowering the price, which they appear to have done with the introduction of the 7970. The next phone I want to get is a Cisco 7920 WiFi... although once again, they're on the exy side, I'm sure they'll also be well worth it. Unfortunately they aren't available in Australia yet, hopefully not too far off. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Thursday, 5 February 2004 12:15 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 quick dial hehe ya I have to admit they are very featureful. :P Asterisk is still a baby i'm sure sip phones will get better with time. But you do have to admit that the cisco 7960's are damn good phones. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Do you Linux softphone..
The link to the download site for the softphone is: http://www.lipz4.com/lipz4.htm On Wed, Feb 04, 2004 at 08:57:00AM +, WipeOut wrote: An article I came across this morning.. http://www.itnews.com.au/storycontent.asp?ID=12Art_ID=18128 -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Boards falling out...
Usually the cards seat pretty well. Do you have a green or blue TDM40B card? Mark On Wed, 4 Feb 2004, Greg Kedrovsky wrote: I have a TDM40B, 4-port fxs card. Each port seems to have it's own little board on the fxs card. Each little board is not sodered in, but rather hangs (I have a vertical case for the server) on what I would call jumper pins (sorry, I'm not a profession geek, just a wannabe). One of my little boards, over time, slides off those jumper pins. I just noticed it this morning. I had to power down, seat it, and power up again. That's a pain. We did, though, have an earthquake this morning. That may have shaken things loose a bit. But, it wasn't much to speak of (long, but not strong). Has anyone else experienced this problem? What could I do to solve it (seat the little card a little more permanently)? Thanks ahead of time. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] talking clock
Thanks for your reply Brian. I am able to get only the hour and minute but not the seconds. I need seconds also, any suggestions? Regards Deepak - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 02:23 PM Subject: Re: [Asterisk-Users] talking clock SayUnixTime will do that just give it the format you want. SayUnixTime([unixtime][|[timezone][|format]]) unixtime: time, in seconds since Jan 1, 1970. May be negative. defaults to now. timezone: timezone, see /usr/share/zoneinfo for a list. defaults to machine default. format: a format the time is to be said in. See voicemail.conf. defaults to ABdY 'digits/at' IMp Returns 0 or -1 on hangup. bkw On Wed, 4 Feb 2004, Deepakumar JV wrote: Hello I am looking for a AGI application that can say the current time with seconds, but i don't need the day/year. Has anyone got this already? Thanks in advance Deepak ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Boards falling out...
On Wed, 4 Feb 2004, Greg Kedrovsky wrote: Has anyone else experienced this problem? What could I do to solve it (seat the little card a little more permanently)? Automotive parts places sell products like lok-tite (a thread locker compound for mechanical fasteners). A drop or two of that, placed over the seam in the connector (after it's fully seated!) would probably hold it in place. This is the same kind of stuff you see on the screws that hold PCBs in their place in nearly everything. It seems to do well keeping those from rattling loose, so it would probably work for your card too. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Boards falling out...
On Wed, Feb 04, 2004 at 08:42:04AM -0600, Mark Spencer wrote: Usually the cards seat pretty well. Do you have a green or blue TDM40B card? Blue. -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] talking clock
Search bugs.digium.com their was a patch for seconds but I don't think it was applied yet bkw On Wed, 4 Feb 2004, Deepakumar JV wrote: Thanks for your reply Brian. I am able to get only the hour and minute but not the seconds. I need seconds also, any suggestions? Regards Deepak - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 02:23 PM Subject: Re: [Asterisk-Users] talking clock SayUnixTime will do that just give it the format you want. SayUnixTime([unixtime][|[timezone][|format]]) unixtime: time, in seconds since Jan 1, 1970. May be negative. defaults to now. timezone: timezone, see /usr/share/zoneinfo for a list. defaults to machine default. format: a format the time is to be said in. See voicemail.conf. defaults to ABdY 'digits/at' IMp Returns 0 or -1 on hangup. bkw On Wed, 4 Feb 2004, Deepakumar JV wrote: Hello I am looking for a AGI application that can say the current time with seconds, but i don't need the day/year. Has anyone got this already? Thanks in advance Deepak ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cepstral TTS Code
Feedback for the list. I compiled Andy's code. Installation went well (except for me misspellng something in the dialplan) with no problems. The Application works great. Will run down Brian's and give it a try too. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Whats wrong with dialplan?
I am having problems with my dial plan, please help me locate the problem:
In the following dialplan, I am not able to press 8 to get to voicemail
main while the 3000 mailbox unavailable message is being read in the
background.
What am I doing wrong?
[globals]
;physical-phones
p1 = SIP/p3000
p2 = SIP/p3001
p3 = SIP/p3002
p4 = some other physical phone
;lines
line1 = Zap/1
[voicemail access]
;Extension 8 to get to voicmail:
exten = 8,1,VoicemailMain
[no match]
exten = _.,1,Playback(sorry-no-match)
exten = _.,2,Hangup
[extensions]
;ext3000:
exten = 3000,1,Dial(${p1},10,tr)
exten = 3000,2,Answer
exten = 3000,3,Background,vm/3000/unavail
exten = 3000,4,Voicemail,3000
exten = 3000,5,Hangup
;If Busy:
exten = 3000,102,Background,vm/3000/unavail
exten = 3000,103,Goto,4
[well-road]
;includes
include = voicmail access
include = extensions
include = no match
exten = h,1,Hangup
[default]
exten = s,1,Goto(well-road,3000,1)
Thanks for any help
Regards
Chris
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Re: [Asterisk-Users] Boards falling out...
Has anyone else experienced this problem? What could I do to solve it (seat the little card a little more permanently)? Haven't experienced it but I would think that a small bead of silicone sealant would hold things in place. As the stuff cures it will release acetic acid but so long as you're not forcing the goop into the connectors (rather just around it) you should be fine. They do make RTV silicone sealant that does not release this acid as it cures; look for Neutral RTV sealant but as I have said, I've *never* run into problems with the normal stuff and holding down electronics so long as you're not going out of your way to get it into the connections. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Boards falling out...
On Wed, Feb 04, 2004 at 07:47:08AM -0700, Greg Hill wrote: Automotive parts places sell products like lok-tite (a thread locker compound for mechanical fasteners). Thanks. I'll give that a try. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Whats wrong with dialplan?
Hi,
-Original Message-
In the following dialplan, I am not able to press 8 to get to
voicemail main while the 3000 mailbox unavailable message is
being read in the background.
What am I doing wrong?
You need to put the exten = 8,... Line in the same context as extensions,
otherwise it won't work like that.
During a Background play, you can access extensions in your _current_
context.
[voicemail access]
;Extension 8 to get to voicmail:
exten = 8,1,VoicemailMain
[no match]
exten = _.,1,Playback(sorry-no-match)
exten = _.,2,Hangup
[extensions]
;ext3000:
exten = 3000,1,Dial(${p1},10,tr)
exten = 3000,2,Answer
exten = 3000,3,Background,vm/3000/unavail exten =
3000,4,Voicemail,3000 exten = 3000,5,Hangup ;If Busy:
exten = 3000,102,Background,vm/3000/unavail
exten = 3000,103,Goto,4
Florian
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Re: [Asterisk-Users] Cepstral TTS Code
[EMAIL PROTECTED] wrote: Feedback for the list. I compiled Andy's code. Installation went well (except for me misspellng something in the dialplan) with no problems. The Application works great. Will run down Brian's and give it a try too. Hope you can do us a HOWTO. Cepstral would be a major win IMO compared to Festival. I use Frank, and even though he sounds a bit effete, my customers love him. I currently generate static GSMs and then play them. Being able to do it inside asterisk would be way cool. BTW what is Andy's code? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Whats wrong with dialplan?
Do you evern include the [well-road] context?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Lee
Sent: Wednesday, February 04, 2004 9:11 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Whats wrong with dialplan?
I am having problems with my dial plan, please help me locate the
problem:
In the following dialplan, I am not able to press 8 to get to voicemail
main while the 3000 mailbox unavailable message is being read in the
background.
What am I doing wrong?
[globals]
;physical-phones
p1 = SIP/p3000
p2 = SIP/p3001
p3 = SIP/p3002
p4 = some other physical phone
;lines
line1 = Zap/1
[voicemail access]
;Extension 8 to get to voicmail:
exten = 8,1,VoicemailMain
[no match]
exten = _.,1,Playback(sorry-no-match)
exten = _.,2,Hangup
[extensions]
;ext3000:
exten = 3000,1,Dial(${p1},10,tr)
exten = 3000,2,Answer
exten = 3000,3,Background,vm/3000/unavail
exten = 3000,4,Voicemail,3000
exten = 3000,5,Hangup
;If Busy:
exten = 3000,102,Background,vm/3000/unavail
exten = 3000,103,Goto,4
[well-road]
;includes
include = voicmail access
include = extensions
include = no match
exten = h,1,Hangup
[default]
exten = s,1,Goto(well-road,3000,1)
Thanks for any help
Regards
Chris
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[Asterisk-Users] IAX2 Problem
Hi, I have setup * from IAX2 and for the client the IAXphone (sokol). When I try to call an demo-extension there is a notice: [114696]: chan_iax2.c:4341 socket_read: rejected connect attempt from my_ip Any idea ? Regards, MarinBlu Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it!
[Asterisk-Users] Asterisk 0.7.2
Asterisk 0.7.2 is now released and contains lots and lots of bug fixes from the bug tracker. Highly recommended for people running 0.7.1. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Code Hosting...
At 10:18 AM +0100 2/4/04, Andy Powell wrote: lo, Is there a single central location for code and applications other than CVS? I'm talking about code that can't/wont be included in CVS for various reasons? Does the wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has done some updates too) but apparently this will never make it into CVS (since the engine is not GPL)... Seems to make sense to have a central location for this type of 'outlaw' code... The bug tracker is useless for this sort of thing but there seem to be a number of bits of code like this in there moo Andy Isn't this what the asterisk-addons directory was created for? This is where the MySQL code was relegated after it became legally unfavorable to put it in the CVS main branches. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Whats wrong with dialplan?
Chris Lee wrote: I am having problems with my dial plan, please help me locate the problem: In the following dialplan, I am not able to press 8 to get to voicemail main while the 3000 mailbox unavailable message is being read in the background. What am I doing wrong? [well-road] ;includes include = voicmail access include = extensions include = no match voicemail is misspelled - would that do it? Bob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ParkAndAnnounce - Get Parking Extension
Is there a way to get the extension that was used to park the call using the ParkAndAnnounce command into a variable? Or a variable that is set? I would like to create an application that allows the person the call is being announce to be able to accept the call (by pressing 1) or send the call to voicemail by pressing (2), but I need to know what parked extension the call is on. If the only way is to modify the source I'm comfortable with that. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Code Hosting...
On Wed, 4 Feb 2004, John Todd wrote: At 10:18 AM +0100 2/4/04, Andy Powell wrote: lo, Is there a single central location for code and applications other than CVS? I'm talking about code that can't/wont be included in CVS for various reasons? Does the wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has done some updates too) but apparently this will never make it into CVS (since the engine is not GPL)... Seems to make sense to have a central location for this type of 'outlaw' code... The bug tracker is useless for this sort of thing but there seem to be a number of bits of code like this in there moo Andy Isn't this what the asterisk-addons directory was created for? This is where the MySQL code was relegated after it became legally unfavorable to put it in the CVS main branches. I think we need some control on what makes it into the asterisk-addons directory. Someone needs to do quility control so that not just anyone can put code there. Maybe make a bug marshel in charge of that area (BKW ?). Let them control it make sure that it is current and working code. MIchael ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail auth failure
When I access voicemail remotely, from a gsm phone say, some extra characters get inserted in my dtmf tones: when I type , * understands 88f8f8 (it always seems to be 'f'): -- Incorrect password '88f8f8' for user '2130' (context = any) And the 'f' always starts after the second digit. Might it be related to this warning message? Feb 4 16:40:59 WARNING[622613]: res_adsi.c:234__adsi_transmit_messages: Unknown ADSI response 'f' This is on debian unstable with 7.1 packages. -- Every day is a gift, that's why the present is so named ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 0.7.2
On Wed, Feb 04, 2004 at 09:21:28AM -0600, Mark Spencer wrote: Asterisk 0.7.2 is now released and contains lots and lots of bug fixes from the bug tracker. Highly recommended for people running 0.7.1. Great, I was going to grab the latest CVS version after business hours today anyway :) Cheers, Steve -- Steve Foy| http://www.unite.net UNITE Solutions | Tel: 028 9077 7338 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: iax, trunking, etc.
Chris Clifton wrote: The majority of sip to pstn gateway providers (vonage, voicepulse, and others) appear to be setup for a one line only type of set up. Their web sites seem to be heavily geared for these one line setups. Anyone willing to comment on what type of pricing plans these providers offer when using iax2 trunking or other methods with asterisk to send multiple (and possibly simultaneous) calls through their gateways ? Voicepulse told me that there was no additional charge to enable trunking. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Boards falling out...
I have a TDM40B, 4-port fxs card. Each port seems to have it's own little board on the fxs card. Each little board is not sodered in, but rather hangs (I have a vertical case for the server) on what I would call jumper pins (sorry, I'm not a profession geek, just a wannabe). One of my little boards, over time, slides off those jumper pins. I just noticed it this morning. I had to power down, seat it, and power up again. That's a pain. We did, though, have an earthquake this morning. That may have shaken things loose a bit. But, it wasn't much to speak of (long, but not strong). Has anyone else experienced this problem? What could I do to solve it (seat the little card a little more permanently)? I haven't looked at an FXS card recently... But when we use to build computers for shipping over seas the IDE card would unseat from the VESA Local Bus during shipping... (God I am probably dating myself with that one.) We would use zip ties (flexible plastic ratchet type of fastener) to hold the cards in place. Plastic == non-conductive and holds like a son of a gun. Not sure if this would work with the FXS card, but just throwing out an alternative to adhesives that might break down due to heat or other environmental issues. We used hot glue for a time, but the zip ties just worked much better. Tom Walsh Network Administrator http://www.ala.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Re: iax, trunking, etc.
Nufone is setup for it and it works great On Wed, 4 Feb 2004, Stephen R. Besch wrote: Chris Clifton wrote: The majority of sip to pstn gateway providers (vonage, voicepulse, and others) appear to be setup for a one line only type of set up. Their web sites seem to be heavily geared for these one line setups. Anyone willing to comment on what type of pricing plans these providers offer when using iax2 trunking or other methods with asterisk to send multiple (and possibly simultaneous) calls through their gateways ? Voicepulse told me that there was no additional charge to enable trunking. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] GS and NAT
Matteo Brancaleoni wrote: hi. I've gs working under NAT, simply put nat=yes into sip.conf section if *, then enable nat into the gs, without any stun server. If I do that (which I already have tested) * will try to initiate rtp with dest IP eq the inside adress ie 192.168.0.160. BTW nat=yes or nat=1 is treated likewise? :( Matteo. Il mar, 2004-02-03 alle 21:17, Tomas Prybil ha scritto: Hi all. Is it at all possible to have a GS B101 NATed with firmware 1.0.4.40? I've tried both STUN and not STUN. The odds seems best with stun because the phone registers with right ip adress. When the connection is made * sends rtp packets to the right destination AND port, but the phone doesn't accept the packets. Should I burn my D-LINK 604 or upgrade the GS? /t ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Cepstral TTS Code
Andy's code and my code are the same code basically. I cleaned up a few things and added the noanswer option. Other than that Andy did all of the hard work. bkw On Wed, 4 Feb 2004, Brian Capouch wrote: [EMAIL PROTECTED] wrote: Feedback for the list. I compiled Andy's code. Installation went well (except for me misspellng something in the dialplan) with no problems. The Application works great. Will run down Brian's and give it a try too. Hope you can do us a HOWTO. Cepstral would be a major win IMO compared to Festival. I use Frank, and even though he sounds a bit effete, my customers love him. I currently generate static GSMs and then play them. Being able to do it inside asterisk would be way cool. BTW what is Andy's code? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Re: iax, trunking, etc.
Voicepulse told me that there was no additional charge to enable trunking. GASP SWOON!!! You received a response out of voicepulse? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Whats wrong with dialplan?
Bob Klepfer wrote: voicemail is misspelled - would that do it? Yup that fixed it, thanks for all the help Regards Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Do you Linux softphone..
Would be great if I could actually download it. It looks nice, does it work? On Wed, 2004-02-04 at 08:37, Walker Haddock wrote: The link to the download site for the softphone is: http://www.lipz4.com/lipz4.htm On Wed, Feb 04, 2004 at 08:57:00AM +, WipeOut wrote: An article I came across this morning.. http://www.itnews.com.au/storycontent.asp?ID=12Art_ID=18128 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie Question. Is asterisk right for my scenario?
Hi, Please excuse me if my question seems too simplistic. I have been reading the mailing list for some time and I am still a bit confused. Here is the scenario that I would need to achieve and am wondering if asterisk is the correct software to use. (h323) (h323/SIP) (h323) pstn---cisco--Asterisk??-cisco---pstn | | | -sip phone I have an existing h323 structure doing h323 pstn termination and would like add sip to part of the structure, also at the same time would like asterik to act as a softswitch to store dial plans and make routing decisions. Asterisk at the same time will do h323/SIP translation. My question, can Asterisk do all these? Or am I totally off? Any comments are welcome. Many Thanks to all. Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 MGCP dialtone problems, part 1 [long]
Hi all, I've been trying on and off again for several months to get my 7960 (MGCP 5.3) working with * with no success. As you know, working MGCP configs for non-ATA Ciscos seem to be very hard to come by. I'm not shooting for the moon here, just trying to get dialtone at the moment. The problem I'd like to focus on today: I only get dialtone when I go off-hook (via the Speaker button, if it matters) maybe once every 3 tries. If it fails, or after I've successfully gotten the dialtone once, the phone will not get it again until it has been power-cycled. This is a failed attempt below. Config files are at the bottom. See my next message for a successful attempt. = initial registration: = MGCP read: RSIP 1 [EMAIL PROTECTED] MGCP 1.0 RM: restart from 192.168.144.225:2427MGCP read: RSIP 1 [EMAIL PROTECTED] MGCP 1.0 RM: restart from 192.168.144.225:2427Verb: 'RSIP', Identifier: '1', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Handling request 'RSIP' on [EMAIL PROTECTED] Transmitting: 200 1 OK to 192.168.144.225:2427 -- MGCP Asked to indicate tone: on [EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 2 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 R: hd(N) to 192.168.144.225:2427 MGCP read: 200 2 OK from 192.168.144.225:2427MGCP read: 200 2 OK from 192.168.144.225:2427Verb: '200', Identifier: '2', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines == going offhook: == MGCP read: NTFY 2 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 O: hd from 192.168.144.225:2427MGCP read: NTFY 2 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 O: hd from 192.168.144.225:2427Verb: 'NTFY', Identifier: '2', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on [EMAIL PROTECTED] Transmitting: 200 2 OK to 192.168.144.225:2427 -- Creating connection for [EMAIL PROTECTED] in cxmode: sendrecv callid: 5d77f7f876d91892 We're at 192.168.144.100 port 16348 Answering with capability 4 Answering with capability 8 Posting Request: CRCX 3 [EMAIL PROTECTED] MGCP 1.0 C: 5d77f7f876d91892 L: p:20, a:PCMU, a:PCMA M: sendrecv X: 76d91892 v=0 o=root 16680 16680 IN IP4 192.168.144.100 s=session c=IN IP4 192.168.144.100 t=0 0 m=audio 16348 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 to 192.168.144.225:2427 -- MGCP Asked to indicate tone: dl on [EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 4 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 R: hu(N), hf(N), D/[0-9#*](N) S: dl to 192.168.144.225:2427 -- MGCP mgcp_new(MGCP/[EMAIL PROTECTED]) created in state: Down MGCP read: 200 3 OK I: 0 v=0 o=- 7960 7960 IN IP4 192.168.144.225 s=MGCP Call c=IN IP4 192.168.144.225 t=0 0 m=audio 26536 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 from 192.168.144.225:2427MGCP read: 200 3 OK I: 0 v=0 o=- 7960 7960 IN IP4 192.168.144.225 s=MGCP Call c=IN IP4 192.168.144.225 t=0 0 m=audio 26536 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 from 192.168.144.225:2427Verb: '200', Identifier: '3', Endpoint: 'OK', Version: '(null)' 2 headers, 7 lines Capabilities: us - 12, them - 260, combined - 4 Non-codec capabilities: us - 1, them - 0, combined - 0 MGCP read: 200 4 OK from 192.168.144.225:2427MGCP read: 200 4 OK from 192.168.144.225:2427Verb: '200', Identifier: '4', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: NTFY 3 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 O: hu from 192.168.144.225:2427MGCP read: NTFY 3 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 O: hu from 192.168.144.225:2427Verb: 'NTFY', Identifier: '3', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on [EMAIL PROTECTED] Transmitting: 200 3 OK to 192.168.144.225:2427 -- Modified [EMAIL PROTECTED] with new mode: recvonly on callid: 5d77f7f876d91892 Posting Request: MDCX 5 [EMAIL PROTECTED] MGCP 1.0 C: 5d77f7f876d91892 M: recvonly X: 76d91892 I: 0 R: L/hd(N) to 192.168.144.225:2427 -- MGCP mgcp_hangup(MGCP/[EMAIL PROTECTED]) on [EMAIL PROTECTED] -- Delete connection 0 [EMAIL PROTECTED] with new mode: recvonly on callid: 5d77f7f876d91892 Posting Request: DLCX 6 [EMAIL PROTECTED] MGCP 1.0 C: 5d77f7f876d91892 X: 76d91892 I: 0 to 192.168.144.225:2427 -- MGCP Asked to indicate tone: on [EMAIL PROTECTED] in cxmode: recvonly Posting Request: RQNT 7 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 R: hd(N) to 192.168.144.225:2427 -- MGCP mgcp_hangup(MGCP/[EMAIL PROTECTED]) on [EMAIL PROTECTED] set vmwi(-) -- MGCP Asked to indicate tone: vmwi(-) on [EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 8 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 R: hd(N) S: vmwi(-) to 192.168.144.225:2427 MGCP read: 200 5 OK I: 0 v=0 o=- 7960 7960 IN IP4 192.168.144.225 s=MGCP Call c=IN IP4 192.168.144.225 t=0 0 m=audio 26536 RTP/AVP 0 a=rtpmap:0 PCMU/8000 from 192.168.144.225:2427MGCP read: 200 5 OK I: 0 v=0 o=- 7960 7960 IN IP4 192.168.144.225 s=MGCP Call c=IN IP4 192.168.144.225 t=0 0 m=audio
[Asterisk-Users] 7960 MGCP dialtone problems, part 2 [long]
Hi all, I've been trying on and off again for several months to get my 7960 (MGCP 5.3) working with * with no success. As you know, working MGCP configs for non-ATA Ciscos seem to be very hard to come by. I'm not shooting for the moon here, just trying to get dialtone at the moment. The problem I'd like to focus on today: I only get dialtone when I go off-hook (via the Speaker button, if it matters) maybe once every 3 tries. If it fails, or after I've successfully gotten the dialtone once, the phone will not get it again until it has been power-cycled. This is a successful attempt below. Config files are at the bottom. = initial registration: = MGCP read: RSIP 1 [EMAIL PROTECTED] MGCP 1.0 RM: restart from 192.168.144.225:2427MGCP read: RSIP 1 [EMAIL PROTECTED] MGCP 1.0 RM: restart from 192.168.144.225:2427Verb: 'RSIP', Identifier: '1', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Handling request 'RSIP' on [EMAIL PROTECTED] Transmitting: 200 1 OK to 192.168.144.225:2427 -- MGCP Asked to indicate tone: on [EMAIL PROTECTED] in cxmode: inactive Posting Request: RQNT 11 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 R: hd(N) to 192.168.144.225:2427 MGCP read: 200 11 OK from 192.168.144.225:2427MGCP read: 200 11 OK from 192.168.144.225:2427Verb: '200', Identifier: '11', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines === offhook, dialtone, dial 1000, hangup; === MGCP read: NTFY 2 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 O: hd from 192.168.144.225:2427MGCP read: NTFY 2 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 O: hd from 192.168.144.225:2427Verb: 'NTFY', Identifier: '2', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on [EMAIL PROTECTED] Transmitting: 200 2 OK to 192.168.144.225:2427 -- Modified [EMAIL PROTECTED] with new mode: sendrecv on callid: 7e0ef17b76d91892 Posting Request: MDCX 12 [EMAIL PROTECTED] MGCP 1.0 C: 7e0ef17b76d91892 M: sendrecv X: 76d91892 I: 0 R: L/hu(N),L/hf(N),D/[0-9#*](N) to 192.168.144.225:2427 -- MGCP Asked to indicate tone: dl on [EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 13 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 R: hu(N), hf(N), D/[0-9#*](N) S: dl to 192.168.144.225:2427 -- MGCP mgcp_new(MGCP/[EMAIL PROTECTED]) created in state: Down MGCP read: 515 12 NO CONNECTION FOR CONNECTION ID from 192.168.144.225:2427MGCP read: 515 12 NO CONNECTION FOR CONNECTION ID from 192.168.144.225:2427Verb: '515', Identifier: '12', Endpoint: 'NO', Version: 'CONNECTION FOR' 1 headers, 0 lines MGCP read: 200 13 OK from 192.168.144.225:2427MGCP read: 200 13 OK from 192.168.144.225:2427Verb: '200', Identifier: '13', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: NTFY 3 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 O: D/1 from 192.168.144.225:2427MGCP read: NTFY 3 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 O: D/1 from 192.168.144.225:2427Verb: 'NTFY', Identifier: '3', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on [EMAIL PROTECTED] Transmitting: 200 3 OK to 192.168.144.225:2427 -- MGCP Asked to indicate tone: dl on [EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 14 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 R: hu(N), hf(N), D/[0-9#*](N) S: dl to 192.168.144.225:2427 -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/[EMAIL PROTECTED] -- MGCP Asked to indicate tone: on [EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 15 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 R: hu(N), hf(N), D/[0-9#*](N) to 192.168.144.225:2427 -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/[EMAIL PROTECTED] -- MGCP Asked to indicate tone: on [EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 16 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 R: hu(N), hf(N), D/[0-9#*](N) to 192.168.144.225:2427 MGCP read: 200 14 OK from 192.168.144.225:2427MGCP read: 200 14 OK from 192.168.144.225:2427Verb: '200', Identifier: '14', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: 200 15 OK from 192.168.144.225:2427MGCP read: 200 15 OK from 192.168.144.225:2427Verb: '200', Identifier: '15', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: 200 16 OK from 192.168.144.225:2427MGCP read: 200 16 OK from 192.168.144.225:2427Verb: '200', Identifier: '16', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: NTFY 4 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 O: D/0 from 192.168.144.225:2427MGCP read: NTFY 4 [EMAIL PROTECTED] MGCP 1.0 X: 76d91892 O: D/0 from 192.168.144.225:2427Verb: 'NTFY', Identifier: '4', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on [EMAIL PROTECTED] Transmitting: 200 4 OK to 192.168.144.225:2427 -- MGCP Asked to indicate tone: on [EMAIL PROTECTED] in cxmode:
[Asterisk-Users] Re: Anyone used a Grandstream ATA286 with Asterisk
MLS Drop for SysAdmin wrote: an associate of mine sent me an email of the slick sheet on this one. I understand that mentioning this vendor has resulted in some flamethrowing on the list, and I do not want to cause trouble - just looking for some info. Thanks! Sam Z I have one in service. It has no serious problems. The one issue we have had is that it adds an additional 2-wire to 4-wire hybrid and therefore adds an echo source that the remote end hears (varies depending upon what you plug into it) - and, I suspect that, because of the somewhat longer packet delay, the echo from this source is not well corrected by the * echocanceller. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] talking clock
On Wednesday 04 February 2004 08:58, Brian West wrote: Search bugs.digium.com their was a patch for seconds but I don't think it was applied yet It was applied; it's just not part of the default. 'S' is the digit for speaking seconds. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] MeetMe questions
The Meetme needs to monitor DTMF and be able to trigger an AGI. and When the Meetme room is emptied it needs to be notified back to * so you can trigger a clean-up event or what not. that is what i would like to see. :) There really aren't any. Once you're in a conference, you can only exit the conference, if you entered with option p specified in the dialplan, by pressing a #. Otherwise, the only way to exit a conference is to hangup. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Code Hosting...
Andy Powell wrote: lo, Is there a single central location for code and applications other than CVS? I'm talking about code that can't/wont be included in CVS for various reasons? Does the wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has done some updates too) but apparently this will never make it into CVS (since the engine is not GPL)... Seems to make sense to have a central location for this type of 'outlaw' code... The bug tracker is useless for this sort of thing but there seem to be a number of bits of code like this in there I have setup http://www.sf.net/projects/asterisk and store various things in that CVS Repository. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Newbie Question. Is asterisk right for my scenario?
Hi, Please excuse me if my question seems too simplistic. I have been reading the mailing list for some time and I am still a bit confused. Here is the scenario that I would need to achieve and am wondering if asterisk is the correct software to use. (h323) (h323/SIP) (h323) pstn---cisco--Asterisk??-cisco---pstn | | | -sip phone I have an existing h323 structure doing h323 pstn termination and would like add sip to part of the structure, also at the same time would like asterik to act as a softswitch to store dial plans and make routing decisions. Asterisk at the same time will do h323/SIP translation. My question, can Asterisk do all these? Or am I totally off? Quite simply, yes. Asterisk is a softswitch more than anything else. And it can take an incoming call on any of its available protocols (PSTN, IAX, SIP, H323, plus many more) and route it back out any of them. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Boards falling out...
Greg Kedrovsky wrote: I have a TDM40B, 4-port fxs card. Each port seems to have it's own little board on the fxs card. Each little board is not sodered in, but rather hangs (I have a vertical case for the server) on what I would call jumper pins (sorry, I'm not a profession geek, just a wannabe). One of my little boards, over time, slides off those jumper pins. I just noticed it this morning. I had to power down, seat it, and power up again. That's a pain. We did, though, have an earthquake this morning. That may have shaken things loose a bit. But, it wasn't much to speak of (long, but not strong). Has anyone else experienced this problem? What could I do to solve it (seat the little card a little more permanently)? Thanks ahead of time. -Greg There are three issues here, relating to the other posts on this topic. Don't use loktite. Loktite is what is called an anerobic adhesive. Specifically, it is catalyzed by contact with metal in the absence of oxygen. As such, it will only cure (in the absence of some other chemical activator) only down inside the pin sockets, holding them together. The rest will stay uncured and spread all over other stuff. This may essentially make them a single use contact. The silicone is a good bet. The acid referred to is the acetic acid (i.e., vinegar) released when the monomers in the RTV goo cross react to form the silicone. Once the cure is complete, there is no acid production and what was produced diffuses away. Mild acids are not terribly corrosive to most metals, and not at all corrosive to gold. The types of RTV that don't produce acid may actually produce alkali (ammonia), which is far more corrosive, but also diffuses away readily. Nevertheless, I would stick to the stuff that smells like vinegar. Finally, I have found that the best approach is the simplest, when it works. If you can get one of those nylon tie-wraps around the daughter card in such a way as to hold it in place, this is the best - and most reversible approach. Sometimes, there are appropriate holes in the motherboard, othertimes the ty-wrap can be snaked around under the connector - however, don't run it under any other type of component. I have even drilled holes in 2-layer circuit boards, but I would not advise this unless you really, really, really know what you are doing. Finally, if the female side of pin sockets are loose enough to let the dayghter cards fall out, they may also be the source of noisy, intermittent connections. Sockets of allmost all kinds are notorious for this kind of thing. I can't tell you how many times I have repaired a flakey circuit board by removing the sockets and soldering in all the (formerly) socketed chips. The square pin spring contacts in those connectors are only designed for a few insertion/removal cycles. If that is the case, you should get a good repair tech. to replace them. Good luck and hang in there. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail volume level?
Are there any parameters that can bump voicemail volume up just a little? I can't seem to find anything but thought I'd ask the list. Normal pstn calls via x100p's are reasonable (very little echo) and running -0.5 db on xmt rcv within zapata. But using the exact same path (pstn via x100p) the voicemail gsm audio seems a little bit low. options? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Boards falling out...
Jon Pounder wrote: nail polish and liquid-paper work fine for this sort of stuff. Too brittle. Adhesion to metal and many plastics is marginal. Fine for places where there is no shock (of the physical kind). If this is earthquake territory, stick to the silicone or ty-wraps. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interrupted musiconhold sound when silence suppression is enabled
Hi, I am a new player of the Asterisk. I have a strage problem with musiconhold feature. Can anyone give some clues what might be the problem? A description of the problem is as follows: 1. Call from Cisco ATA 186 without silence suppression, when I push the "hold" button at the Cisco 7960 IP phone, the music plays just fine. 2. Call from Cisco ATA 186 with silence suppression enable, when I push the "hold" button of phone, the music is annoying, it cannot play smoothly. Sometimes, itplays well for a while, then there is a pause, then plays again, then pause,... Certainly, an obvious solution is to disable silence suppression, however, this is unpracticable, because, sometimes, you might have no control of the remote side. Thanks in advance. George Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it!
[Asterisk-Users] New Search engine for the list - Final resting place
I've found a home for my new search engine of the Asterisk users mailing list. Thanks to Linkx in the Netherlands (http://www.linkx.net) for hosting this. There are a number of search resources for this list. This is another. This one is a little different however, you can do a fuzzy phrase search if desired and you can restrict the search within specific periods, from certain users, to require certain words in the subject and other features. It's also pretty fast. It's subscribed to the list and will be updated daily. Have fun. - Kim Hendrikse ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cepstral TTS Code
Hi Brian, Andy's code and my code are the same code basically. I cleaned up a few things and added the noanswer option. Other than that Andy did all of the hard work. is cepstral a special tts-api, or does this mean, we can use every windows(tm) tts-engine on the market...? Even ATT Natural Voices...? Can we allready test this app, or is this a closed source thingy...? Greez Andreas _ Surf the net and talk on the phone with Xtra Jetstream @ http://www.xtra.co.nz/products/0,,5803,00.html ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: Compiling while * is running
Stephen R. Besch wrote: I just fetched today's cvs (1/30/04 11:10:31). Compiles/installs on my test machine (ASUS A7V, 900 MHZ). However, If I try to compile on my production machine (Elite K7S5A, 2.4GHz, 512MB) while * is running the zaptel and asterisk compiles seg fault. I am assuming that they will compile correctly if I bring down * and rmmod the zaptel driver. 0.7.1 compiled and is now running. Is there a way to safely compile while * is running, so that I can minimize down time of the server? Here's the update on the seg fault problems. After bad memory was suggested, I checked to see what memory the tech who assembled the machine had used. I don't know why, but the module was not in the recommended list for the MOBO at Crucial.com. I ordered one of the recommended modules. It arrived this morning. I installed it and all of the seg faults are gone. I've learned my lesson. Next time, I will assemble my own machine! Thanks again for all the help. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: New Search engine for the list - Whoops!
Whoops! I forgot to mention it's location :) Here you go: http://asterisk.linkx.net/cgi-bin/asterisk - Kim Hendrikse I've found a home for my new search engine of the Asterisk users mailing list. Thanks to Linkx in the Netherlands (http://www.linkx.net) for hosting this. There are a number of search resources for this list. This is another. This one is a little different however, you can do a fuzzy phrase search if desired and you can restrict the search within specific periods, from certain users, to require certain words in the subject and other features. It's also pretty fast. It's subscribed to the list and will be updated daily. Have fun. - Kim Hendrikse ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] 7960 MGCP dialtone problems, part 1 [long]
Hiya, I've been trying on and off again for several months to get my 7960 (MGCP 5.3) working with * with no success. As you know, working MGCP configs for non-ATA Ciscos seem to be very hard to come by. I'm not shooting for the moon here, just trying to get dialtone at the moment. The problem I'd like to focus on today: I only get dialtone when I go off-hook (via the Speaker button, if it matters) maybe once every 3 tries. If it fails, or after I've successfully gotten the dialtone once, the phone will not get it again until it has been power-cycled. I could not get 5.3 to work, but 6.1 seems to work. Basic Phone that is, i don't get *any* buttons on the phone, i guess this is a problem with CARD.XML, the only version on CCO ist for version 3.0 (!). If anyone has a working one, please post it on the list ;-) Regards, aa _ Surf the net and talk on the phone with Xtra Jetstream @ http://www.xtra.co.nz/products/0,,5803,00.html ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip flow diagram?
Does anyone have a high level flow diagram showing acceptable sip messages exchanges? For exampe: Source Dest Invite - -Trying Ok - I'm specifically trying to debug an issue with various hangups, prior to call completion, after call completion, calling vs called party hold, etc, and getting rather confused watching the various packets flowing between sip devices with a sniffer (and no reference document). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Code Hosting...
Isn't this what the asterisk-addons directory was created for? This is where the MySQL code was relegated after it became legally unfavorable to put it in the CVS main branches. JT The code in question was actively denied entry into CVS (asterisk core or addons) Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] ParkAndAnnounce - Get Parking Extension
Yup, the only way to do that is to modify the source. The task of setting a channel variable to the parked slot is simple; however, because the parkandannounce application is placing the call to the callee to announce, it would need to have the logic of handling the press 1 or 2 inside itself, rather than in the call plan. Not a bad feature, but definitely an edit to the source. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlton J. O'Riley Sent: Wednesday, February 04, 2004 10:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ParkAndAnnounce - Get Parking Extension Is there a way to get the extension that was used to park the call using the ParkAndAnnounce command into a variable? Or a variable that is set? I would like to create an application that allows the person the call is being announce to be able to accept the call (by pressing 1) or send the call to voicemail by pressing (2), but I need to know what parked extension the call is on. If the only way is to modify the source I'm comfortable with that. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Sip flow diagram?
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_programm ing_reference_guide_book09186a0080080221.html Stephen Dolloff DLS Internet Services 847-854-4799 x256 [EMAIL PROTECTED] -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 11:45 AM To: Asterisk-a-users-list Subject: [Asterisk-Users] Sip flow diagram? Does anyone have a high level flow diagram showing acceptable sip messages exchanges? For exampe: Source Dest Invite - -Trying Ok - I'm specifically trying to debug an issue with various hangups, prior to call completion, after call completion, calling vs called party hold, etc, and getting rather confused watching the various packets flowing between sip devices with a sniffer (and no reference document). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Sip flow diagram?
Try RFC 3261 http://www.faqs.org/rfcs/rfc3261.html Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, February 04, 2004 12:45 PM To: Asterisk-a-users-list Subject: [Asterisk-Users] Sip flow diagram? Does anyone have a high level flow diagram showing acceptable sip messages exchanges? For exampe: Source Dest Invite - -Trying Ok - I'm specifically trying to debug an issue with various hangups, prior to call completion, after call completion, calling vs called party hold, etc, and getting rather confused watching the various packets flowing between sip devices with a sniffer (and no reference document). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Notice: This e-mail message, together with any attachments, contains information of Merck Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp Dohme or MSD and in Japan as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Do you Linux softphone..
On Wed, 2004-02-04 at 17:36, Chris Tooley wrote: Would be great if I could actually download it. It looks nice, does it work? Well after a few false starts, I've got as far as:- it registers with * and can make calls, but for some reason doesn't accept calls. I'll have to check configs. On Mandrake and maybe others it looks for it's modules in the wrong place but this fix works:- LD_LIBRARY_PATH=/usr/local/zultys/kylix3/bin softphone I've found it easier to edit the config file directly than use the menu system all the time. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Re: Boards falling out...
On Wed, Feb 04, 2004 at 12:17:46PM -0500, Stephen R. Besch wrote: nail polish and liquid-paper work fine for this sort of stuff. Too brittle. Adhesion to metal and many plastics is marginal. Fine for places where there is no shock (of the physical kind). If this is earthquake territory, stick to the silicone or ty-wraps. That's good to know. Thanks. Yes, it's earthquake territory. Nothing major, be we get bounced around regularly. Volcanos - active and inactive. The country (Costa Rica) is one big volcano. -Greg -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Re: Boards falling out...
On Wed, 2004-02-04 at 12:12, Colin Anderson wrote: I have used hot glue for many years with no problems. Decent adhesion, but can be picked off if ness. I showed this to a systems integrator that had problems with shipping PC's upside down and boards would become unseated. He used this on thousands of systems and the problem was eliminated. Hmm, sounds like something the Digium resellers could do. I can see the commercial now cheesy announcer voice Is your access to insert local emergency number important to you. Do you want to not have to go looking for your tools in the middle of an emergency. For the low low cost of $29.99, we will throw in a hot glue gun and 3 sticks of glue to solve all those earthquake related failures. /cheesy announcer voice -Original Message- From: Stephen R. Besch [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 04, 2004 10:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Boards falling out... Greg Kedrovsky wrote: I have a TDM40B, 4-port fxs card. Each port seems to have it's own little board on the fxs card. Each little board is not sodered in, but rather hangs (I have a vertical case for the server) on what I would call jumper pins (sorry, I'm not a profession geek, just a wannabe). One of my little boards, over time, slides off those jumper pins. I just noticed it this morning. I had to power down, seat it, and power up again. That's a pain. We did, though, have an earthquake this morning. That may have shaken things loose a bit. But, it wasn't much to speak of (long, but not strong). Has anyone else experienced this problem? What could I do to solve it (seat the little card a little more permanently)? Thanks ahead of time. -Greg There are three issues here, relating to the other posts on this topic. Don't use loktite. Loktite is what is called an anerobic adhesive. Specifically, it is catalyzed by contact with metal in the absence of oxygen. As such, it will only cure (in the absence of some other chemical activator) only down inside the pin sockets, holding them together. The rest will stay uncured and spread all over other stuff. This may essentially make them a single use contact. The silicone is a good bet. The acid referred to is the acetic acid (i.e., vinegar) released when the monomers in the RTV goo cross react to form the silicone. Once the cure is complete, there is no acid production and what was produced diffuses away. Mild acids are not terribly corrosive to most metals, and not at all corrosive to gold. The types of RTV that don't produce acid may actually produce alkali (ammonia), which is far more corrosive, but also diffuses away readily. Nevertheless, I would stick to the stuff that smells like vinegar. Finally, I have found that the best approach is the simplest, when it works. If you can get one of those nylon tie-wraps around the daughter card in such a way as to hold it in place, this is the best - and most reversible approach. Sometimes, there are appropriate holes in the motherboard, othertimes the ty-wrap can be snaked around under the connector - however, don't run it under any other type of component. I have even drilled holes in 2-layer circuit boards, but I would not advise this unless you really, really, really know what you are doing. Finally, if the female side of pin sockets are loose enough to let the dayghter cards fall out, they may also be the source of noisy, intermittent connections. Sockets of allmost all kinds are notorious for this kind of thing. I can't tell you how many times I have repaired a flakey circuit board by removing the sockets and soldering in all the (formerly) socketed chips. The square pin spring contacts in those connectors are only designed for a few insertion/removal cycles. If that is the case, you should get a good repair tech. to replace them. Good luck and hang in there. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 bug in 6.1 evident in Asterisk
Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Sip flow diagram?
Rich Adamson wrote: Does anyone have a high level flow diagram showing acceptable sip messages exchanges? For exampe: Source Dest Invite - -Trying Ok - I'm specifically trying to debug an issue with various hangups, prior to call completion, after call completion, calling vs called party hold, etc, and getting rather confused watching the various packets flowing between sip devices with a sniffer (and no reference document). Rich It may be a little verbose, but you can find it in the rfc 3261 as a start. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cepstral TTS Code
No it uses the linux theta libs and header files. bkw On Wed, 4 Feb 2004, Andreas Anderson wrote: Hi Brian, Andy's code and my code are the same code basically. I cleaned up a few things and added the noanswer option. Other than that Andy did all of the hard work. is cepstral a special tts-api, or does this mean, we can use every windows(tm) tts-engine on the market...? Even ATT Natural Voices...? Can we allready test this app, or is this a closed source thingy...? Greez Andreas _ Surf the net and talk on the phone with Xtra Jetstream @ http://www.xtra.co.nz/products/0,,5803,00.html ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Cepstral TTS Code
http://asterisk.bkw.org/other/cepstral.tar.gz bkw On Wed, 4 Feb 2004, Brian Capouch wrote: I'm prolly showing my ignorance here, but where *is* this code? I've done a search at the bugs site and it came up dry. It's not in the CVS contrib tree. Don't know where else to look. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 0.7.2
would be a good idea to put it on the changelog, i see its there but it doesnt really inform nothing. Miguel Cavazos On Wed, 2004-02-04 at 15:21, Mark Spencer wrote: Asterisk 0.7.2 is now released and contains lots and lots of bug fixes from the bug tracker. Highly recommended for people running 0.7.1. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Sip flow diagram?
You can find some examples here: http://www.iptel.org/info/players/ietf/callflows/ Enjoy reading... SIP is like poetry! Christian -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, February 04, 2004 6:45 PM To: Asterisk-a-users-list Subject: [Asterisk-Users] Sip flow diagram? Does anyone have a high level flow diagram showing acceptable sip messages exchanges? For exampe: Source Dest Invite - -Trying Ok - I'm specifically trying to debug an issue with various hangups, prior to call completion, after call completion, calling vs called party hold, etc, and getting rather confused watching the various packets flowing between sip devices with a sniffer (and no reference document). Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Code Hosting...
On Wed, Feb 04, 2004 at 10:18:10AM +0100, Andy Powell wrote: but apparently this will never make it into CVS (since the engine is not GPL)... GPL code is not allowed in the Digium CVS repository. Only split-licensed code is. /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asuscom HiSax based ISDN BRI card - one way latency
Hi all, I have configured Asterisk server with Asuscom ISDN card and 1 port TDM10B card. ISDN card is based on HiSax chipset and it runs with hisax kernel module. When call from 'zap/1' phone (or any SIP client) to PSTN is connected next behaviour occurs: - PSTN station hears 'zap/1' with latency (about 1 sec) - 'zap/1' hears PSTN station without latency I've spent a lot of time by tracing the cause. A few minutes ago I've find something really strange. When kernel module is loaded and asterisk run (e.g. asterisk -c) it works fine. Once you connect to running asterisk server by 'asterisk -r' then one way latency starts! The only way to get a rid of latency is to stop asterisk, remove hisax module and start all again. I've tested Asuscom ISDN card (HiSax: HFC-PCI card manufacturer: Asuscom/Askey) and Eicon Diva (HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2). Asuscom works as described. With Eicon card there is no latency at all. It doesn't matter if remote connection to asterisk was made or not. Any ideas why remote connection to asterisk server confuses Asuscom ISDN card? Why Eicon Diva is not affected? It uses the same chipset and kernel module... My configuration: PC: Athlon XP 2200+, 512MB RAM, Debian-stable (Woody), Kernel 2.4.22 Asterisk - updating continuously from CVS - I get the same behaviour over two months modem.conf is as follows (nothing special): [interfaces] driver=i4l type=autodetect dialtype=tone mode=immediate stripmsd=0 group=1 msn=xx context=isdn device = /dev/ttyI0 device = /dev/ttyI1 Thank you, poorman ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 bug in 6.1 evident in Asterisk
Does the first line, backup and emergency proxy go to the * box on the same wire? Malcolm and I figured out the 7960's freak smooth out if the asterisk server isn't on the same subnet his phones kept rebooting over and over and over till we took them off the switch they were on and move them to the one with the aterisk server. bkw On Wed, 4 Feb 2004, John Todd wrote: Yes and no. The Cisco phone is on a NAT network that is quite distant from one of the Asterisk servers, but on the same wire as the other. Three lines go to the remote *, and three lines remain local on the network to the other * server. I'm running CVS as of this morning on both servers. Strangely, today the phone hasn't locked up or rebooted, though now I am getting one or two of the lines failing to REGISTER - they're simply not sending out a request, according to the network dump. sigh JT At 7:43 AM -0600 2/4/04, Brian West wrote: Question.. is the 7960 on the same subnet as your asterisk server? I have a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running 6.1 and has 12 days of uptime. bkw On Wed, 4 Feb 2004, John Todd wrote: So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to the point where it needs to be unplugged, due to software errors. This is a first. My suspicions are that this bug in Asterisk is causing the lockups: http://bugs.digium.com/bug_view_page.php?bug_id=889 It seems unusual to me that a low volume of bogus SIP messages should lock up the 7960, but that seems to be the case. It seems this only happens on my 7960 that I have completely full of extensions (all six line buttons are lit, two of them are auto-answer.) I think this is one bug tickling another bug; bad messages from * are killing the 7960. I'd like anyone else with experiences with this type of failure with Asterisk to give me a shout; I'm going to report this to Cisco somehow, but don't have enough evidence. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio code registration
I am pulling my hair out trying to get an AUdiocodes MP-108 FXO gateway to register. I set it up like some of my other phones but keep getting errors. Here is the SIP debug. Any pointers would be appreciated. 10 headers, 0 lines Message is NOTIFY Sip read: REGISTER sip:216.139.32.179 SIP/2.0 Via: SIP/2.0/UDP 216.139.30.77;branch=z9hG4bKacgKAyfjp From: sip:[EMAIL PROTECTED];tag=1c25733 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 208177 REGISTER Expires: 180 Contact: sip:[EMAIL PROTECTED];user=phone;expires=180 Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 216.139.30.77 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 216.139.30.77;branch=z9hG4bKacgKAyfjp From: sip:[EMAIL PROTECTED];tag=1c25733 To: sip:[EMAIL PROTECTED];tag=as4fe5fd3a Call-ID: [EMAIL PROTECTED] CSeq: 208177 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
