Re: [Asterisk-Users] Cisco 7960 SIP 7.4
On Mon, 14 Mar 2005 08:06:20 -0800, Scott Laird [EMAIL PROTECTED] wrote: I don't see any major changes in the release notes--mostly small bug fixes. They fixed some DHCP and NTP problems, as well as a 802.1x problem with some of their switches. There were a couple SIP protocol fixes in there too, plus a spelling fix. Has anyone else upgraded to 7.4 and found that the date time no longer appears on the phone? Ie: The phone doesn't appear to be grabbing the date time off the NTP server on my network, it worked alright on 7.3 (except for the time drift) but now they seem to have fixed the drift by no longer displaying time nor date. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP 7.4
On Sun, 27 Mar 2005 20:06:39 +1000, Chris Lee [EMAIL PROTECTED] wrote: Ie: The phone doesn't appear to be grabbing the date time off the NTP server on my network, it worked alright on 7.3 (except for the time drift) but now they seem to have fixed the drift by no longer displaying time nor date. Problem sorted... something is wrong with my local NTP server, I've now changed my config to get the time off my ISP's NTP server and it's working fine (note to self: make sure you use the IP address for the server and not a DNS name). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk@home scary log
Colin Anderson wrote: The hack came in through ssh. IMO, your best defence is an extremely strong root password; I am often mortified by looking at my logs and seeing all of the login attempts through SSH. OT: I am not up on Linux script-kiddie type tools, but I assume that there is a script of some sort that automates SSH probes. Can anyone suggest a good counter i.e. honeypot or throttling logon attempts. Yes, I know I can google it, but I'd rather hear the opinion of real Linux experts rather than the experts at About.com. Most scripts use port 22 as it would be too big a task to scan for ssh on all ports, so I run my ssh server way above port 1024. This has, touch wood, prevented any unusual activity in the last few months. Chris. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to set up a server compatible with Windows apps ?
DEMAINE Benoit-Pierre wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 ~ I would like to : set up a server on Linux on which my friends can connect with msn or netmeeting, suporting at least sound conferance, and optionally video, but I dont want asterisk server to lock up the sound card; and then, I want to be able to connect that server with a free Linux tool; I had a look at http://www.voip-info.org/wiki-Asterisk but did not find any help; ~ I tried 'sudo asterisk -f -v' , and got for the user : $ asterisk -f -r -R Unable to connect to remote asterisk Try sudo asterisk -vc to start asterisk, then you may see why it is not starting. 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] GSM phones, bluetooth and general happiness
Joe Antkowiak wrote: There are quite a number of positive (for end users) implications of doing this too... just think about all those cell providers that offer unlimited mobile to mobile calls, and then all those unlimited LD packages from landline and voip providers. This has huge potential for people who use their cell phones alot... Then make sure the channel allows you to: pool devices, set free minutes on each device,and have preference for devices with remaining free minutes, thus sharing the calls between my phone and that of my wife. An IAX/(sip if it must) softphone with appropriate extensions to work with bluetooth devices could provide a solution without having a Bluetooth dongle in the PABX. 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] Silently Wait for DTMF Input
[EMAIL PROTECTED] wrote: Hello! I would like to call a number (e.g.35), and when i press a secret code (12345), it should jump to my voicebox menu. On this page http://www.voip-info.org/wiki-Asterisk+cmd+background i found something about Silently Wait for DTMF Input. In my case it wouldn`t be silence. It woudl just play the away message. Now how can i include such a secret code to my background funktion? I am looking for something _like_ this: exten = s,1,Background(away,12345,voicebox_35) If someone presses 12345 whilest the away message is playing, it jumps to the voicebox_35 context. Know what i mean?! A quick link to the needed docu or some tips/examples would be great! Thanks, Mario What you want is an extension 12345 in the same context as the extension 35 that will be used when you dial 12345 while background is playing the message. in the 12345 extension you do the normal 'voicemailmain' with the skip password feature if you want to go straight there. Just have a good think about security though, before you leave your voicemail open to every one. 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] DTMF information?
Steve Underwood wrote: Chris Lee wrote: I am looking at building an IVR product with a few interesting features and need some more information about how asterisk and VoIP work and what I can get from them. As far as I can tell when I use ISDN/GSM telephone networks the DTMF information travels as data representing 'start tone' and 'stop tone' for each button pressed, it is then generated at the other end if an audio representation is required. I am interested to know if I can get access to these events 'start tone' and 'stop tone' through the dialplan or an AGI or by acting as a VoIP device. Or of course if I am completely off track and should give up now. I am looking to get the length of time a button was held down rather than that it was pressed. Thanks for any help ISDN never does this. GSM only does this between the handset and the base-station. You only see DTMF tones from outside the GSM network itself. For fancy IVRs, beware that the timing of DTMF from a GSM handset has nothing to do with the timing of the user's keypresses. Because the base-station generates the tones, it controls their timing, and always generates rather long slow pulses of DTMF tones. Regards, Steve Thanks for the info, I will have to re-think my plans. With GSM the tone lasts until the user releases the key, this I have tested GSM to GSM and GSM to Land Line (pots). I assume then that the ISDN carries DTMF as audio and this must be decoded by something? hardware or software depending on the card in the PC. I have noticed with analogue modems that the hardware decoders provide a 'DTMF tone received' indicator and then ignore the length of the tone, Does any one know if tone length is acquired by any of the asterisk drivers? and retained/passed on within asterisk? It seems like throwing away information, which could be useful, if the length of the key press is ignored and only the fact it was pressed is retained. I dont know what the standards say but I have noticed that all phones I have tested (except DECT phones) send a continuous tone until the button is released. 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] Free WWT (WorldWideTelco): Utopia, or just a matter of organization?
Jon Radel wrote: Marconi Rivello wrote: In US, local calls are free. So it wouldn't be a problem to make such a network to get rid of long distance calls. But in other countries (like here in Brazil) local calls are charged. So there could be some king of billing (without commercial purposes, just to pay for the costs), or something... Well... actually... There are ever fewer people left in the U.S. who have free local outbound calls. Lots of people have enough calls or minutes included in the base monthly charge to cover their normal calling every month, so it sort of looks like the calls are free. However, if their phone is suddenly in use 18 hours / day by people all over the world their bill might jump painfully. For example, if I exceed my monthly allowance on my residential POTS lines with Verizon, I pay 9.6 cents per local call. However, if your phone company in the U.S. ever noticed that you were billing or sharing your line, they'd probably make you get a more expensive business line, and that's if they were feeling really, really nice. (A Local Exchange Carrier (LEC) not to be named here once shut down one of our DS-3s without notice because we thought we were wholesaling the capacity, in part because we were carrying VOIP across it. Their lawyers turned green, and sent people with fancy titles to apologize, after they realized what the business office had done, but it, and lesser episodes, show that the LECs in the U.S. are, as a general rule, quite freaked out about all this VOIP and free phone call stuff. You are messing with their revenue, you know.) All of which pales besides how you'll feel the first time the police drop by to discuss a phone call placed from your phone that played a key role in some nasty crime. The chances are you'd have some very long and very painful discussions with police officers who wouldn't know an RTP packet if it punched them in the nose about how a) it wasn't you, b) it wasn't somebody who came over to your house to use your phone, c) you might be able to figure out what country the call was from if they let you go home and look at your logs, and d) they should believe you and not the technicians from the phone company. So, I'd second other recommendations here that do this if you must for a few friends you trust, but think really, really hard before you open something like this to the public. --Jon Radel A few friends who you trust can get quite large when the system is like the PGP web of trust, if the 'project' were to require that a trusted member were required to 'sponsor' a new trusted member at risk of their membership then the users could possibly grow quite nicely. Add to that a required Asterisk configuration if you want to connect e.g. 'Must have these restrictions enabled (only calls originating from controlled sources may travel via trunks)', 'must provide 2 trunk connections (one to sponsor and one to group voted destination)','Must publish Asterisk config or relevant part to group or at least connected parties' etc. It could be quite an interesting community. Of course if it were a true web of trust environment with any member able to use anothers directly without routing through others then login names and passwords would be useless and pgp keys would have to be used so that revoked members could be excluded without massive maintenance, and new members added by similar means. 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
[Asterisk-Users] DTMF information?
I am looking at building an IVR product with a few interesting features and need some more information about how asterisk and VoIP work and what I can get from them. As far as I can tell when I use ISDN/GSM telephone networks the DTMF information travels as data representing 'start tone' and 'stop tone' for each button pressed, it is then generated at the other end if an audio representation is required. I am interested to know if I can get access to these events 'start tone' and 'stop tone' through the dialplan or an AGI or by acting as a VoIP device. Or of course if I am completely off track and should give up now. I am looking to get the length of time a button was held down rather than that it was pressed. Thanks for any help 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] GSM to BRI ISDN Gateway
Miroslav Nachev wrote: Hi, I am looking for GSM to BRI ISDN Gateway. Any help? I was also looking for such things nd came across these guys: http://www.2n.cz/export they have a product or two for GSM and here is the one I found most likely to work for me (two GSM sim cards providing two ISDN channels on a BRI line): http://www.2n.cz/uploads/2/PAGES/C379.HTML But I still have to get hold of one for testing, the local supplier is moving offices and as such can not help me out in the short term. 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] Can PSTN CallerID be fowarded to a SIP phone extension?
James Freire wrote: Hi All, I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID from the PSTN to the SIP phone that is setup as an extension? We have a Voice menu setup for incomming calls and I would like to recieve the caller ID of the calls we are recieving after the incomming caller reaches their final destination. Thanks! -James Yes, I can't remember the exact places to put what, but you can set calerid to asprovided or something like that in Zapata.conf or zaptel.conf or something. Have a look on www.voip-info.org, use search on CLI and the zaptel config file. almost everything you need is in here somewhere. 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] just a few newbie questions
[EMAIL PROTECTED] wrote: Hello List! I just read an article about asterisk, and i would like to ask a few questions to see if i understood the principle right. Reciving Calls: --- - To be able to recive calls, i need to have an VoIP-Provider. No, You can use one but you could also install a hardware PSTN/ISDN card for receiving and making calls via your Telco - I need a Static IP, so that the VoIP-Provider can redirect the call to my office/asterisk server.- asterisk will then forward the call to my phone If you want to receive calls via a VoIP provider it makes things easier if your address is static. Otherwise you need to register your address with the providers system each time it changes, I dont think this is an automated feature of asterisk. Making Calls: --- - To be able to make calls, i need to have an VoIP-Provider. See Above a Hardware interface can do the trick - I need an account from a VoIP-Provider If you want them to take your traffic. - A call from a Softphone/Office will be made to the asterisk server, which will then forward it to the VoIP Provider, which will then transfer it to the normal telephone network. Yes, unless you have a hardware interface then it goes SoftPhone - Asterisk - Telco Correct to far? Thanks, Mario 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
[Asterisk-Users] Off Toppic-ish Telephony question
I have a IVR system which is causing me some difficulties and am after some help to see if it is the source of the problems or just a victim of the cause. The system is running windows 2K and it uses the the CAPI interface to receive calls through a FRITZ ISDN card. Now the trouble I have is with DTMF recognition when the caller is in noisy environments. If the noise exceeds a certain (as yet unknown) threshold then the system is not responding to DTMF. Now my question is the following: The users are all using Nokia GSM hand sets. As my limited knowledge recalls GSM sends the DTMF as signaling data and the ISDN should receive the DTMF without requiring operation of the audio channel. Am I correct? If I am correct why do I have this problem; Is it due to the nokia trying to turn the complex noise into DTMF and overriding the keyboard input? Is it due to the CAPI software (at the server) getting audio channel and trying to get DTMF out of that and thus overriding DTMF from phone keyboard (If so I thought that the GSM used lower quality codec so how is DTMF arriving via audio channel?) This all leads me to the situation of if it is the CAPI end that is causing me the problem maybe I can use asterisk and prevent audio channel DTMF collection (is this simple) but if it is elsewhere what can I do? (change handsets?) Thanks for any insight on this Regards 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] X101P on a UK BT line ---- txgain issue
I do get echo, lots of it, I am waiting until the new patch they are all on about on the list gets into a stable release, then I will upgrade and see if that does the trick. I am told that some of the echo may be to do with a mismatch in the impedance with the BT line. I had an adsl problem a while back and the engineer fixing it said I had a constant, 50 something ohm, loop condition if the X100p was plugged in. If we could fiddle the impedance matching maybe it would fix things a bit. But I am no phone engineer so am not sure on any of this. Chris. Chris Stenton wrote: Chris, Do you get echo issues? If not could you let us have your config and which echo canceller you use. Thanks Chris On Thu, 2004-06-24 at 20:40, Chris Lee wrote: Chris Stenton wrote: I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with an ordinary phone. Yes I have this too (BT LINE), I upped mine to 10 in order that other people could hear me without a problem. I can hear them fine. 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] X101P on a UK BT line ---- txgain issue
Chris Stenton wrote: I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with an ordinary phone. Yes I have this too (BT LINE), I upped mine to 10 in order that other people could hear me without a problem. I can hear them fine. 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] Multiple DTMF digits on 7960
B. J. Bomar wrote: Hello all. We have an asterisk system set up, and we are seeing a lot of multiple DTMF digits being read by asterisk. In digging through the archives the only answer I have seen is to put in the statement relaxdtmf=yes in the zapata.conf file. Since we are not using any zapata devices, I have tried to put that statement in my sip.conf file to no avail. Any help would be appreciated as my end users are starting to get highly annoyed. Thanks, B. J. If you are using G711 ALAW/ULAW as your codec and you have the DTMF set to sip info or RFC it is possible that the hardware you are using is duplicating it on the audio channel and that asterisk is decoding this. This will result in the data arriving twice, once via sound the other via sip. I cant remember any specific solutions, but I think the grandstream needs to have its DTMF set to sip info to prevent extra DTMF happening. There was a bit on this in here a long time ago so have another trawl through the archives. Good luck 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] Cost of IP Phones, or Isn't It Just Software?
SNIP On the other hand... Go take a look at all of the ~$100 wireless router/firewall/print server/gateway boxes on the market, and you'll see one thing that almost all of them have in common: they all run Linux. Most of them are even based on the same small number of tools; things like busybox and uclibc. If you want to see cheap, powerful VoIP phones, think about what they really need in terms of software, and then set out to write it and license it so the phone companies can incorporate it into their products. I'm kind of amazed that FXS ports aren't standard on medium-end home routers right now; they'd probably only add $5-10 to the cost of the router, *IF* they had the software and felt like the demand was there. My Draytek ADSL 2600v comes with two FXS ports ___ 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] Draytek Vigor 2600Vi as SIP client on Asterisk
Michael Hamann wrote: Hi Everybody, as a relative newby I´m just trying to get a Draytek Vigor Router (2600Vi) connected to my Asterisk System (CVS-05/31/04). With X-Lite and a Cisco Phone it is no problem, but the Vigor seems to have some problems with Asterisk. The first thing ist when I do a sip show peers on the console I get: 4002/4002172.16.183.37 (D) 255.255.255.255 5060 Unmonitored 4001/4001172.16.183.37 (D) 255.255.255.255 5060 Unmonitored What does this status unmonitored mean? With my softphone the entry looks like: 6275/6275172.16.181.49 (D) 255.255.255.255 5060 OK (8 ms) The next thing is that when I try to call one of the vigors SIP Ports via X-Lite I see the following message in the debug console: Jun 18 10:09:54 NOTICE[131081]: chan_sip.c:5150 handle_response: Dunno anything about a 0 Unkown status code response from SIP/4001-b2fc No call is signalled to the phone. The other way, my X-Lite rings but the connection is hung up the moment I accept the call. The Draytek support says that the Vigor does not support SIP Reinvite and that I should try to disable it in my PBX system. So I changed my sip.conf to: [4001] type=friend username=4001 secret=4001 mailbox=2000 canreinvite=no context=default host=dynamic But it still does not work. Does anybody has this combination working and could send me his config files? Or any other ideas? best regards from germany 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 I had this working once, now I have a grandstream so it is no longer needed. It is vital that you get the latest version of the firmware for the vigor as previous versions do not work with the sip server on the lan ports only on the other side of the ADSL line. The reason for this is the sip packets always originated from the ADSL address instead of the internal address which is the one you want to be using if you have an internal server. Next I used a settup a bit like this: Vigor: VOIP SETUP SIP Related Functions SIP: SIP Port 5060 Registrar asterisk.mydomain.com (or an IP address) Port1: Name: p1 Password: (I did not use one) Expiry Time: 10 mins VOIP Setuip CODEC/RTP etc: Codecs: G.711MU Packet Size: 20ms DTMF: OutBand Payload Type 101 RTP: Take the default ports Asterisk: Sip.conf: [general] port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind to context=in-sip ; Default for incoming calls callerid=Call 909090 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm maxexpirey=1800 defaultexpirey=600 tos=throughput [p1] type=friend host=dynamic user=p1 ;secret= dtmfmode=rfc2833 [EMAIL PROTECTED] callerid=p1 3002 qualify=yes context=home hope this helps 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] Asterisk-Users List Etiquette
Kevin Walsh wrote: Steven Critchfield [EMAIL PROTECTED] wrote: You forgot to add in how awful it is when people post using HTML and then override font sizes or assume blue is an appropriate font color for their message. While I know some people don't like it when I turn my attention to them, if it takes me even one more button press to be able to read your mail, it isn't likely to be interesting to me to even bother helping you with your problem. Since the majority of unix users understand how each of us tweak our environment to be the most productive for us, we don't like it when you take liberties with our settings. He also forgot to mention how awful it is when people lazily top-post instead of taking the time to format their followups correctly. This is especially true when trying to follow a thread found in the archives. I fully agree with your anti-HTML comments, by the way. I think you will find that about half the people out there disagree with this sentiment (a guess based on the number of top and bottom posters I have seen) so no matter how often you ask it is not likely to change things much. Top posting is what a lot of people are very comfortable with. It also has the advantage in lists that when you step through a thread the answer to the last item is ready for you to read. So If you bottom post you make life harder for the thread reader but if you top post you make life harder for those that get a long mail out of the archives.Who should we favor? Don't ask why I am bottom posting, I have no good reason, it just so happens that I am. I don't like HTML either but a lot of people don't know they can switch it off or that it even exists (its a word processor isn't it?). Getting offended by these personal preferences just leads to that etiquette problem, the god ol flame war. Or at least heated debate that will never be won with so many advocates for each side, that the lists become quite full of top/bottom html/text arguments. Please don't bring these subjects into things it just makes people with other views upset. ___ 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] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
usedcanon wrote:
Quite simple really,
You could do the following assuming your area code is 0207 (london !)
exten = 9NXXNXXX,1, Dial(SIP/0207${EXTEN})
Umar.
The London code is 020 the 7 or the 8 is part of the local number now.
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[Asterisk-Users] GSM to ISDN or TAPI
Hi I am in the UK and am looking for a device that will allow me to connect two sim cards (read wireless lines) to either the port on the back of my fritz card or any other connection direct to the PC that provides a usable telephony interface. I will even plug two devices into a windows box and have that do ISDN to ISDN if required. All I want is two GSM lines that look like voice modems to the PC and provide full telephony interface, that is DTMF both ways CLI and a few other bits and pieces. I am looking to using asterisk as a remote IVR for looking after some equipment, but land lines are a problem. Any help is much appreciated 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] GSM to ISDN or TAPI
Storer, Darren wrote: Hi Chris, CL All I want is two GSM lines that look like voice modems to CL the PC and provide full telephony interface, that is DTMF CL both ways CLI and a few other bits and pieces. We use the Nokia 22: http://www.nokia.com/nokia/0,,56024,00.html They have worked well providing both telephony applications on remote sites and SMS support for Broadcast work in the UK (serial AT command interface). If you don't mind single band (900 or 1800 MHz GSM) operation there is an older device (Nokia Premicell) that can be sourced cheaply from eBay: http://www.nokia.com/cda1/0,1080,2700,00.html Does the incoming DTMF and voice work over the serial interface with the 22? I had a Nokia 32 for test and could not get it to return DTMF, it has AT commands to generate DTMF and to receive CLI but I could not get it into voice mode or get DTMF out of it. Thanks for your help 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] GSM to ISDN or TAPI
Storer, Darren wrote: Hi Chris, CL Does the incoming DTMF and voice work over the serial CL interface with the 22? I can't help but feel that you are going about this all the wrong way (based upon the limited information you have chosen to share with us). If you need to pass control information from one node to another and you have a pair of Nokia 22s then why not simply send an SMS message? Maybe this is not a good solution for you but until you fill in the gaps it's the best I can come up with. IE. Tell us a bit more about what you are trying to achieve. I am going to have some remote machines which need to have adjustments made to their settings on occasion, the most cost effective and user friendly way I can come up with is a simple IVR system that says press 1 to set limits on flow, press two for flow status report etc. It will use a combination of CLID and pin number to authenticate the engineer doing the config. So what I need is something that can take a pay as you go sim (least cost for line rental) and accepts calls without too many problems. As I said earlier I had a Nokia 32; I plugged it into a windows box with USR voice modems and it would not work at all, it only provides a dial tone when connected to select hardware, certain phones and one or two winmodems, I could not justify the cost of the X100p for testing based on the mixed results. So now I am looking at finding some other method of linking the GSM to the PBX, I like the concept of digital all the way and RS-232 looks like voice modem would be great. Hope that clears up my requirements. Thanks for the help 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
[Asterisk-Users] DTMF X100p to sip GS
I have a GS BT102 which when receiving calls that include DTMF tones only have short clipped beeps. Is asterisk not passing the DTMF info on to the phone or is the tone not bing generated by the phone? I am trying to check that I am getting the DTMF for an application where the length that a key is held down is important so need the SIP device to be sent the start and end of each key press the user makes. Regards 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] Strange CallerId behaviour with SIP
Joost Kraaijeveld wrote:
Hi all,
I want to see the name of the caller (if available) and not the number.
If I call from my IP phone to my software IP phone I see the name of the caller. If I call from the software phone to my IP phone I only see the number, not the name. If I call from IP phone to IP phone I only see the number. If I set the name explicitely using callerid = asrevieved in my sip .conf or if I use SetCallerID(${CALLERIDNAME}) I get aseterisk on my display. IAX calls seem to go OK.
Is this known behaviour or is my configuration wrong? If so, any hints for a sollution?
Could be the phone, some phones dont do Alpha characters, the
Grandstream for example can not display the name, only the number.
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Re: [Asterisk-Users] Asterisk and picoCell GSM Base Stations
The UK is currently not legally set up to allow the use of these devices as there is currently no unlicensed bandwidth in the GSM space. If you want to use one you have to approach the Cellular networks and ask them to install it and connect it directly to their network. You may be able to pay them enough to let you use their space within your building. So using old cell phones is out for the UK until OfCom does something about it, which I hear may be in the pipe line. These things go for about £2000, this is based on an order of 100, and is probably not available through the Cellular network for this price. Regards Chris. Simon Anderson wrote: picoCell/microCell/nanoCell Base stations are low powered GSM nodes designed for indoor or small area GSM coverage. Their transmission power is low enough to meet the legal restrictions of many countries. Here are two examples; http://www.rivanetworks.com/nano/nano.htm http://www.ipaccess.com/ipaccess_pages2/bts2.html Both of these Base Stations have an RJ-45 in order to do Ethernet on the back end; GSM in, IP out. I wonder if anyone has experience interfacing one of these or a similar product with Asterisk? I also wonder if Digium has any plans to supply PCI cards which provide similar functionality? The prospect of recycling old cell phones for use as Asterisk extensions is extremely attractive. Cellular phone ---GSM--- picoCell ---IP--- Asterisk Regards, Simon. ___ 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 SIP Images
Brian Cuthie wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roderick Montgomery Sent: Monday, March 29, 2004 4:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images ... ### ### Hardware != Software ### Cisco IOS Software, phone firmware, etc. is normally bundled with hardware at the time of purchase, because, frankly, the hardware isn't really of much use without software. You may resell the hardware (which, looking at eBay, happens frequently), but the software license DOES NOT transfer from one end user to another. There are only a few exceptions to this rule, such as for business affiliates, mergers, acquisitions, lease buyouts, and outsourcing arrangements. Frankly, this is a horrible policy. It's designed to eliminate the market for used gear so that vendors can force people to buy new equipment. Frankly, anyone with this business model should be ashamed. And anyone buying equipment under such circumstances should beware. The assets they think they're purchasing today have substantially less value than they think since they can't effectively resell them when they're no longer needed. -brian ___ 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 I think it is time to start a Linux on Cisco hardware project, if one does not already exist. ___ 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 CLI in the UK
First, is the lack of UK CLI on the x100P hardware or software related? Secondly, My US Robotics Voice modem does get UK CLI, so could I get UK CLI and the same functionality as the x100p using a USR Modem with *? Has anyone done this? As an aside, has anyone experienced or solved the problem with the x100p producing a loop condition on the PSTN line (it really mucks up my ADSL connection something horrid when it is connected). I think it is due to an impedance mismatch between the card and the network, but have no way of testing these things. (Dont know enough to just get out my meter and start probing without risk of killing my x100p or the POTS Line) I know the Loop condition is there as a kindly BT eng was monitoring the line and asking me to plug things in, when the x100p was plugged in he said something along the lines of: theres your problem, what did you just plug in? It is creating a 36 K Ohm Loop condition Now the router is not the most stable at the best of times but plug in the x100p and the line bounces up and down like there is no tomorrow. 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] message lights and stutter tones
Tim Sailer wrote: On Mon, Mar 08, 2004 at 03:12:52PM -0600, [EMAIL PROTECTED] wrote: Simon, Do the GS phones support stutter tone as-well-as the message light? I'm not Simon, but yes, they do. At least my -100 does. The display backlight flashes, and you get the stutter dialtone. Tim My backlight is flashing and I have the stutter tone, only I dont have any mail waiting, why can I not get the phones to stop doing this? ___ 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] message lights and stutter tones
Simon Chappell wrote: I am no Asterisk mystro (a newbie really) but here is my pennys worth.. I have GS budgettones.. extracts from conf files.. sip.conf [2000] type=friend username=2000 host=dynamic dtmfmode=info [EMAIL PROTECTED] context=sip callerid=2000 secret=password canreinvite=no extensions.conf [sip] ;local extensions exten = 2000,1,Dial(SIP/2000,20,rt) exten = 2000,2,Voicemail(u2000) exten = 2000,102,Voicemail(b2000) exten = 2000,103,Hangup voicemail.conf [local] 2000 = 1999,simon,[EMAIL PROTECTED] Hope that helps Simon Chris Lee wrote: Tim Sailer wrote: On Mon, Mar 08, 2004 at 03:12:52PM -0600, [EMAIL PROTECTED] wrote: Simon, Do the GS phones support stutter tone as-well-as the message light? I'm not Simon, but yes, they do. At least my -100 does. The display backlight flashes, and you get the stutter dialtone. Tim My backlight is flashing and I have the stutter tone, only I dont have any mail waiting, why can I not get the phones to stop doing this? ___ 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 It was the @context on this one too, Thanks for the reply ___ 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] Does it exist - DNS TX record?
When handed a URL type address for telephony, is there a DNS TX record (like MX but for telephone/Video) that could be looked up for an address to use to connect the call? I would like to have a gateway server (probably *) that anyone who knows the email address of a member of staff can use to connect to them with. If the details of this server were in my DNS then anyone trying to call someone at cybericom.co.uk could find the server to make the connection with. 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
[Asterisk-Users] Request for enhancement - IP dependent ports
I am not a programmer so can not implement this, but I think it may be useful. Asterisk configured to listen on multiple IP addresses, Then configure RTP ports for each address independently; So I open 5 ports on one IP and then forward those ports to that IP from my firewall. Then on another IP I can still have hundreds or thousands open for my internal users. That way I can avoid opening many ports to the outside world on my firewall as I dont expect more than a few users a day to use this route. ___ 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] DSL (DMT) goes down when X100 plugged in
Thomas M. Schaefer wrote: Hi all, I have a strange problem. Whenever I plug in the base cord connected to the X100, my DSL service goes down. I DO have a Cisco filter (the one that comes with the product) installed. Has anyone else seen this problem? There was a similar entry in the archives, but it was without a filter. Thanks, Tom Schaefer ___ 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 I have been told that when I plug my X100p into the line I get a 36 Khom loop condition and this may be affecting my ADSL connection (it keeps dropping the line). It may be to do with impedance differences here in the UK, But I know very little about Such things. ___ 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] Confusion with IAX PBX-PBX
I have been trying to set up three * servers to use IAX between them and am a bit lost as to the finer detail of the config files. I have read the wiki and it has not made things better. Here is my problem; I create a section like this on each machines: [othermachine-1] type=friend host=dynamic secret=password trunk=yes qualify=yes context=incoming-1 [othermachine-2] type=friend host=dynamic secret=password trunk=yes qualify=yes context=incoming-2 Now in my extensions.conf I use the link like this: IAX2/othermachine-1 But my problem comes in with the receiving machine, how does it know which machine the link came from without a username of some kind. Or have I completely missed the point of IAX? Please help I am completely lost. Thanks 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
[Asterisk-Users] * nat internet nat sip phone howto
Is there a howto for the situation below? * -- router with nat port forward to * -- router with nat port forward to sip phone -- sip phone ___ 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] Help with Sip call problems - Whats not working?
Wes Marderness wrote: What does your extensions.conf look like? Did you answer() the call first ? The relevent sections of extensions.conf: [voicemail access] ;Extension 8 to get to voicmail: exten = 8,1,Answer exten = 8,2,VoicemailMain [wellingborough-road] ;includes include = emergency include = voicemail access include = external access include = extensions include = no match exten = h,1,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
[Asterisk-Users] two phones one host
I have a sip box with two FXS ports (Draytek 2600v adsl router) I have had very little luck getting the two talking together. For a very short time I did have calls originating on my FXO card routed to the phone working. Phone1/2 on router --- handytone works handytone --- router phone1/2 works Phone1/2 on router --- asterisk broken* asterisk--- router phone1/2 broken** * Can see asterisk receive the number and start following calling plan but no sound comes through, phone eventualy times out. ** Worked for a short time for no apparent reason, now sees phone as bussy, phone does not ring. Router: hostname: gateway-2.cybericom.co.uk phone 1: p3000 phone 2: p3001 IP: 10.10.10.2 Asterisk: hostname: babybell.cybericom.co.uk IP 10.10.10.3 -If I use a phone connected to the router, I see asterisk receive the number and start following dial plan, but I dont hear anything and asterisk retries sending the following packet: Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ors-20101 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=fSd-1369 To: sip:[EMAIL PROTECTED];tag=as07550202 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 232 v=0 o=root 16230 16230 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 13984 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to 10.10.10.2:5060 -If I get a call via my fxo and it is supposed to be routed to a phone on the router I get a 404 not found but it seems asterisk is not asking for a specific phone by name, here is the first packet: -- Executing Dial(Zap/1-1, SIP/p3000|10|tr) in new stack We're at 10.10.10.3 port 13934 Answering with preferred capability 2 Answering with preferred capability 4 Answering with preferred capability 8 Answering with non-codec capability 1 12 headers, 11 lines Reliably Transmitting: INVITE sip:gateway-2.cybericom.co.uk SIP/2.0 Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK04efdbfe From: Cybericom sip:[EMAIL PROTECTED];tag=as179b70c5 To: sip:gateway-2.cybericom.co.uk Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 10 Feb 2004 15:16:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 232 v=0 o=root 16258 16258 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 13934 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 10.10.10.2:5060 -- Called p3000 is there not supposed to be a p3000@ in the sip: line? here is my sip.conf if something is wrong please let me know, thanks: [general] port=5060 ; Port to bind to bindaddr=0.0.0.0; Address to bind to context=in-sip ; Default for incoming calls callerid=Call canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=1800 defaultexpirey=600 tos=throughput [p3000] type=friend host=dynamic user=p3000 ;secret= dtmfmode=rfc2833 mailbox=3000 callerid=Reception 3000 qualify=yes context=wellingborough-road [p3001] type=friend host=dynamic user=p3001 ;secret= dtmfmode=rfc2833 mailbox=3001 callerid=Reception 3001 qualify=yes context=wellingborough-road Router has box for registrar set to 10.10.10.3 place for naming phone set to p3000 and p3001 place for port set to 5060 Thank you for any help Regards 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
[Asterisk-Users] Help with Sip call problems - Whats not working?
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to 10.10.10.2:5060 Retransmitting #2 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 To: sip:[EMAIL PROTECTED];tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to 10.10.10.2:5060 -- Executing BackGround(SIP/p3000-1186, sounds/lots-o-monkeys) in new stack -- Playing 'sounds/lots-o-monkeys' (language 'en') Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 To: sip:[EMAIL PROTECTED];tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to 10.10.10.2:5060 Retransmitting #4 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 To: sip:[EMAIL PROTECTED];tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to 10.10.10.2:5060 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 To: sip:[EMAIL PROTECTED];tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to 10.10.10.2:5060 Feb 9 09:47:15 WARNING[81926]: chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Response) == Spawn extension (wellingborough-road, 8, 2) exited non-zero on 'SIP/p3000-1186' -- Executing Hangup(SIP/p3000-1186, ) in new stack == Spawn extension (wellingborough-road, h, 1) exited non-zero on 'SIP/p3000-1186' set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 10.10.10.2, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK3faaa31d From: sip:[EMAIL PROTECTED];tag=as3bf9fee8 To: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.10.10.2:5060 babybell*CLI Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK3faaa31d From: sip:[EMAIL PROTECTED];tag=as3bf9fee8 To: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE Content-Length: 0 7 headers, 0 lines Message is BYE ### Calls originating at FXO and going to this extension work fine. Calls originating at this extension are a problem. Any help would be great Regards 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:[OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help
On the subject of South Africa What are the laws regarding using the Internet to carry telephone traffic? What are the laws regarding connecting digium kit to Telkom equipment? As I recall they are quite restrictive, have they been eased up a bit? 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
[Asterisk-Users] Asterisk as non root
I followed the wiki instructions: http://www.voip-info.org/wiki-Asterisk+non-root Now I have a working asterisk running as user asterisk. I do however have some problems: 1: I dont have access via asterisk -r 2: The pid file is no longer being updated 3: I want to create a file in init.d so that I can use service start and stop, but need to be able to pass asterisk the gracefully command etc, any ideas welcome. maybe: asterisk -rx stop gracefully etc 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] Asterisk as non root
Tilghman Lesher wrote: Permissions problem. User asterisk needs to have permissions to write the file /var/run/asterisk.ctl 2: The pid file is no longer being updated Again, permissions problem. I was under the impression that changing the line: ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk fixed it so that asterisk.ctl and asterisk.pid got written in the /var/run/asterisk directory, which I gave asterisk ownership of. Am I mistaken here, and if so where do I configure the source so that they get put there, thus getting the rights they need? ___ 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(${p1},10,tr)
exten = 3000,2,Answer
exten = 3000,3,Background,vm/3000/unavail
exten = 3000,4,Voicemail,3000
exten = 3000,5,Hangup
;If Busy:
exten = 3000,102,Background,vm/3000/unavail
exten = 3000,103,Goto,4
[well-road]
;includes
include = voicmail access
include = extensions
include = no match
exten = h,1,Hangup
[default]
exten = s,1,Goto(well-road,3000,1)
Thanks for any help
Regards
Chris
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Re: [Asterisk-Users] Whats wrong with dialplan?
Bob Klepfer wrote: voicemail is misspelled - would that do it? Yup that fixed it, thanks for all the help Regards Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] upgrade problems
})
[pbx_config]
103. Goto(1)
[pbx_config]
[ Context 'voicemail access' created by 'pbx_config' ]
'8' =1. VoicemailMain()
[pbx_config]
[ Context 'macro-Standard-Ext' created by 'pbx_config' ]
's' =1. Dial(${ARG1}|10|tr)
[pbx_config]
2. Answer()
[pbx_config]
3. Background(vm/${MACRO_EXTEN}/unavail)
[pbx_config]
4. Voicemail(3000)
[pbx_config]
5. Hangup()
[pbx_config]
102. Voicemail(b${MACRO_EXTEN})
[pbx_config]
103. Hangup()
[pbx_config]
here is what happens:
Call connects and during unavailable message 8 is pressed which used to
pass me through to voicemailMain and allow access to mail boxes.
now I get hung up and the following messages are displayed:
== Everyone is busy at this time
-- Executing BackGround(Zap/1-1, vm/3000/unavail) in new stack
-- Playing 'vm/3000/unavail' (language 'en')
== CDR updated on Zap/1-1
-- Executing Playback(Zap/1-1, sorry-no-match) in new stack
Feb 3 15:33:58 WARNING[229391]: file.c:446 ast_openstream: File
sorry-no-match does not exist in any format
Feb 3 15:33:58 WARNING[229391]: file.c:734 ast_streamfile: Unable to
open sorry-no-match (format UNKN): No such file or directory
Feb 3 15:33:58 WARNING[229391]: app_playback.c:83 playback_exec:
ast_streamfile failed on Zap/1-1 for sorry-no-match
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (well, 8, 2) exited non-zero on 'Zap/1-1'
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (well, h, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
any input would be great, have I not installed something or is there a
config file which has a new format as of 0.7.1?
please help
Regards
Chris Lee
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[Asterisk-Users] compiling * pipe error
Building * on a machine with a minimal install of Mandrake, worked fine on non minimal install now I get this: bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe If anyone can help me figure out what package I might have missed out when installing mandrake, it would be great. It is possible that the security system is causing a problem, but am not sure here either, any help is greatly appreciated Thanks 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
[Asterisk-Users] Switchboard interface
I am looking to produce a switchboard interface - hopefully web based I needs to: Show the logged in user the CLI of the call they are currently dealing with Show the number of calls in the queue Give a number of options for working with the call transfer put on hold etc. for transfer it must provide a list of users to transfer to. Allow a call to be initiated Can anyone give me ideas on how I might interact with * to get this informations and provide these services. Thanks 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] New sounds also now in CVS
As a sugestion, store the sounds in a soundlib tree, hashed or categorised (boolean (yes, no, true,false, up, down etc.),numbers, caledar(day, date, time etc), state, weather etc) and dont duplicate any sounds then make a sounds tree with virtual categories and sim link to the files needed. This keeps the directory sizes down and allows for sound sets to be built up with all the words they use in them. It also allows sounds to be added as needed rather than requiring all sounds to be part of a distribution. Robert Hajime Lanning wrote: quote who=Tilghman Lesher Although the OS may cache that information, the userland process can take quite some time to process a very full directory. I've had this happen quite a few times with Linux ext2 filesystems, where the fileglob * exceeded bash's limit of 32,768 characters. /bin/ls on those directories took several minutes before the first results were given. I'll additionally comment that the directories I was working with were not normally that full, but was a side effect of a process dumping lots of little files into a directory when something went wrong. On a slight tangent, NT4 had a practical limit of about 300 directory entries before attempting to process the directory became unbearably slow. -Tilghman A couple of things, searching a directory for a specific name tends to be a linear search through the directory (unless the filesystem uses binary trees, like ReiserFS...), ls is a bad example of a command, it is more of a worse case example. ls will read the entire directory, sort it, then do a stat() on every file listed. All of this is done before it formats the output. So, you have to wait until it is all done, before you see the first character output. ___ 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] Newbee question
I am new to asterisk and am wanting to know if it can do some things: in a large/ distributed environment users move about either office to office or branch to branch can they log in and have their virtual extension routed to the one they are on? naturaly this implies the question: if branch servers are used can they ceep track of where a virtual extension is currently attached? ___ 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
