Re: [asterisk-users] IRVs Asterisk example configuration
On Tue, 4 Dec 2007, Vincent wrote: Does someone know why the posts from some users on Usenet are just one long line, with no carriage return? It's called flowed text and defined in RFC 3676. Essentially it lets people with different width screens accomodate paragraphs of text which is line-wrapped on their email clients. See if your email client supports it - maybe it's something you have to enable. Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Text-To-Speech synthesizer--help required
Hello users, Actually i wanted to implement Text-To-Speech engine from cepstral voice using swift application i tried the documentation of doing this and i was unsuccessful at doing this work with asterisk can anybody please help me out finding the solution to installation thanks in advacnce srinivas Antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disturbance noise in the background for digium card
Hi All; I installed one digium card of 2 fxo and 2 fxs, but the following problems related to the voice are happening: 1) Sometimes when I call to the PBX, I hear like modem sound and after little it disapear. 2) There is a disturbance in the background (like the channel radio disturbance that might happen if the frequency was not captured well), and that disturbance appear much more when Asterisk goes via IP Trunk. Is it a configuration issue or it might be a digium card defection so need to be replaced? Any advise? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My AsteriskNo unable to registration
Hi Bie, You have a problem with the postgresql conexion. You are using Realtime? [Dec 5 07:39:59] ERROR[2342] res_config_pgsql.c: Postgresql RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. [Dec 5 07:39:59] DEBUG[2342] res_config_pgsql.c: Postgresql RealTime: Cannot Connect: [Dec 5 07:39:59] WARNING[2342] res_config_pgsql.c: Postgresql RealTime: Couldn't establish connection. Check debug. [Dec 5 07:39:59] DEBUG[2342] res_config_pgsql.c: Postgresql RealTime: Cannot Connect: could not connect to server: Connection refused Is the server running on host 127.0.0.1 and accepting TCP/IP connections on port 5432? Change in your sip.conf put : type=friend failed for 'xxx.xxx.xxx.xxx' - Peer is not supposed to register Tellme something. Guillermo El Miércoles, 5 de Diciembre de 2007 01:48, Newbie escribió: Hi Guillermo, enclosed please find full log file that I got it from /var/log/asterisk please help. Thanks a lot in advance Regards Winanjaya - Original Message - From: Guillermo Rodriguez [EMAIL PROTECTED] To: Newbie [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 04, 2007 5:59 PM Subject: Re: [asterisk-users] My AsteriskNo unable to registration Yes, the log file.. El Martes, 4 de Diciembre de 2007 12:01, Newbie escribió: Hello, could you please advise .. where can I find the trace of asterisk? do you mean log file? Thanks Regards Bie - Original Message - From: Guillermo Rodriguez [EMAIL PROTECTED] To: Newbie [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 04, 2007 5:53 PM Subject: Re: [asterisk-users] My AsteriskNo unable to registration Can you put the trace of asterisk.??' When you call to 988 Thx. Guillermo El Viernes, 30 de Noviembre de 2007 10:17, Newbie escribió: Dear The Expert, I am very new with this, I have installed AsteriskNow, X-Lite as my SoftPhone, I am using SPA-3102. I have 3 extensions, me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below) My problem is, I am unable to call 998, I thought this is registration problem, (because the Linksys screen info said Registration Failed) Could any body please help? Many thanks in advance Regards Bie below is my sip.conf allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm I also had 2 extensions (me at 250 and 998 is my SPA-3102) and my users.conf goes below: [general] fullname=New User userbase=6000 hasvoicemail=yes vmsecret=1234 hassip=yes hasiax=yes hasmanager=no callwaiting=yes threewaycalling=yes callwaitingcallerid=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callgroup=1 pickupgroup=1 host=dynamic localextenlength=0 allow_aliasextns=no allow_an_extns=no hasagent=no hasdirectory=no [250] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=Winanjaya group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=250 secret=1234 threewaycalling=yes vmsecret=1234 zapchan= registeriax=yes registersip=yes canreinvite=no nat=no dtmfmode=rfc2833 disallow=all allow=all type=peer [998] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=MyLine1 group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=999 secret=1234 threewaycalling=yes vmsecret=1234 zapchan= registeriax=yes registersip=yes canreinvite=no nat=no dtmfmode=rfc2833 disallow=all allow=all type=peer [999] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=MyPSTN group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=999 secret=1234 threewaycalling=yes vmsecret=1234 zapchan= registeriax=yes registersip=yes canreinvite=no nat=no dtmfmode=rfc2833 disallow=all allow=all type=peer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Increasing the voice volume from the digium card
Hi All; It is digium analoge card (2 fxo and 2 fxs), so what do I need to use? And where I can find a link for that? Also, is it possible to have a difference voice volumes to be used each for each Trunk or each user? Your kindly help is high appreciated. Regards Bilal bilal ghayyad wrote: Hi List; Anyone knows a method (command) to increase the voice volume at diguim card level? Are you trying to do this at some other level than rxgain and txgain settings in zapata.conf? If so, for the analog cards there are some module parameters for doing so. For digital T1/E1 cards, the only way to do it is with the gain options in zapata.conf. - In zapata.conf you can add rxgain and txgain settings and use ztmonitor to get it set. There are some more details on this on voip-info.org. On Nov 29, 2007 1:49 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Hi All; I have an IP Trunk established between Asterisk and the VoIP service provider, when call from my mobile to the PBX and then enter the destination number to call via the VoIP, I got a connection but the voice level volume need to be increased, I am trying to find if zaptel (diguim card) can increase the volume (if there is any command can do that)? And if that volume level is possible to be applied only for that IP Trunk and not for others. Any Help? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bad behaviour between X-Lite 3.0 and Asterisk
Hello, There is something wrong when using the version 3.0 of X-Lite. When X-Lite sends INVITE, Asterisk replies OK. And it seems, at first sight, that Asterisk ignores the ACK signal sent by X-Lite. There's after a series of Retransmitting of the OK signal, the ACK signals are well received on the Asterisk, but with no effect. And after 6 retransmission, Asterisk hangs up the call. I've looked more in the debug ouputs of Asterisk (sip debug, set verbose 4 and set debug 4), and I've noticed something unusual: at the first OK send by the Asterisk, it receives 2 identical ACK signals from X-Lite. At the first one, Asterisk says SIP TIMER: Cancelling retransmit of packet (reply received) which I think is good. And at the second one, Asterisk says SIP TIMER: Initalizing retransmit timer on packet which I think is not good. The following ACKs in answer to the retransmitted OKs seem to not be matched. I've made the same tests with X-Lite 2.0 and there is no problem, there is no 2 ACKs. Does someone have a similar issue with X-Lite 3.0? Regards, Benoit -- Benoît Mérouze - Telecom Software Developer - IPercom [EMAIL PROTECTED] Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. Benjamin Franklin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
Alex Balashov wrote: I'm sure this has been asked a million times before, but is there an easy wa to have Asterisk register more than one (distinct) contact binding concurrently? The goal is to have two phones register with the same credentials from different locations and consistently and reliably ring on inbound calls, irrespective of their registration intervals and so on. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 A million and one now, check the archives. No you cannot and why would you want to? The device that registers last will ring. Just set the phones up in a ring group or even a ring all queue. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
Philip Prindeville wrote on Tuesday, 04 December 2007 at 11:58 PM: Steve Edwards wrote: On Tue, 4 Dec 2007, Philip Prindeville wrote: I wanted to write a popcorn app for myself, both to learn how to script in Just out of curiosity, what does this have to do with popcorn? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 You used to be able to dial popcorn (767-2676) in any area code (at least prior to 1982) and get the current time. -Philip Actually, this was specific to Northern California (767 prefix). In Southern California, the Time Announcement service has always been in the 853 prefix. The official numbers were 767-1212 and 853-1212, respectively -- though the entire prefix in all area codes of the respective halves of the state were reserved for, and rang to, the Time Announcement service. As of 19th September 2007, ATT discontinued the service due to the unavailability of parts for the 1960s-era Audichron equipment, and declining use of the service. That being said, I would love to have this ability in Asterisk. Perhaps someone has even preserved Jane Barbe's original recordings in a way that they can be recorded for Asterisk. That would be a kick. Sincerely, Trevor Hammonds ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Adtran supervision problems
I am sending a call down a EM wink trunk to a adtran tsu600 channelbank. The extension is setup like so... exten=799179,1,Dial(zap/g2,20,D(9179)) exten=799179,2,Hangup() It should Dial the Adtran and send some DTMF signals to a telephone on an fxs module in the Adtran. Asterisk is seeing the call answered when the T-1 is picked up by the Adtran not when the ringing phone is answered. This means the digits have already been sent by the time the ringing phone is answered. Does anyone have an idea on how to signal this correctly? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP-Realtime and sip reload
Hi, I use SIP-Realtime to store my SIP-users and I keep the informations about the SIP-Providers my Asterisk registers to in sip.conf. I'm running into the following problem. If I set rtcachefriends=yes because I want to use MWI and run a sip reload because I changed something in sip.conf, Asterisk forgets about all registrations of the users which are all unavailable after that. How can I use rtcachefriends=yes to allow MWI (isn't it needed for NAT-keepalive as well?) and don't break everything with a sip reload? thanks for your help Henrik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
SIP wrote: Steve Totaro wrote: Alex Balashov wrote: I'm sure this has been asked a million times before, but is there an easy wa to have Asterisk register more than one (distinct) contact binding concurrently? The goal is to have two phones register with the same credentials from different locations and consistently and reliably ring on inbound calls, irrespective of their registration intervals and so on. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 A million and one now, check the archives. No you cannot and why would you want to? The device that registers last will ring. Just set the phones up in a ring group or even a ring all queue. Thanks, Steve Totaro ___ Why would you WANT to? Well heck, I can think of a lot of reasons -- not the least of which being able to freely log in from anywhere at anytime with multiple phones (the wifi sip phone from the coffee shop, the desk phone at the office, the phone at home, the new phone I just picked up at lunchtime) without having to configure a device entry for each and every one of them and modify extensions.conf to ring multiple devices for each and every phone I add or remove from the list. In short, flexibility. The problem with this question is the way Asterisk thinks of phones to the way many people think of logins. To Asterisk, phones are devices -- separate entities for which there should be an entry each time. To those of us NOT migrating into Asterisk from the traditional PBX world, this is somewhat of a foreign concept. The idea that everywhere we log in from must be a unique device that has to be configured to be allowed to log in is somewhat weird in a world of mobility. In the days of terminals all connecting to a central hub, it made more sense. But in the days of internet cafes, library computers, wi-fi everywhere, etc., it's just not a compatible concept. Who wants to reconfigure his VoIP box every time he goes to a new computer with a new softphone, for instance? So while it may make absolutely PERFECT sense in the realm of Asterisk, as Asterisk is a PBX system and that's how PBX systems think, I'm always surprised at the number of people who simply don't understand why people ask this question. A lot. :) N. Every machine in a in a Windows environment must be configured to join a domain. A user must also be setup in that domain to log in. It is more secure to lock that user into a single login session so that if they are logged in at one machine, they cannot login somewhere else. Think of it like that. Flexibility is not always best practice nor secure. I do not see how internet cafes and wifi have anything to to do with anything. If you go to any of these places with your softphone or wifi phone, they should work. I am not sure how you would expect a computer to just know how to configure itself other than setting up a download site with a provisioning tool. AFAIK, computers cannot read minds yet, nor just configure themselves without human intervention. If you want to be that flexible you can just configure Asterisk to allow you to auto register and use authenticate on dialing or to be really flexible, just leave it wide open until you file to file bankruptcy due to toll fraud. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
Ryan Burke wrote: I just was looking over the app_waitutil.c and am confused you add 500 to tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec + 500) / 1000);? Without having looked at Philips code at all, that looks like he is rounding up? /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use of slin as a codec
Where bandwidth is not an issue but good call quality is, is there any theoretical quality improvement to be had by using slin as the codec over an inter-Asterisk IAX trunk rather than a-law (or u-law in the US). Does anyone know what the slin bandwidth is (compared to 64 kbps a-law). Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adtran supervision problems
Jordan Novak wrote: I am sending a call down a EM wink trunk to a adtran tsu600 channelbank. The extension is setup like so... exten=799179,1,Dial(zap/g2,20,D(9179)) exten=799179,2,Hangup() It should Dial the Adtran and send some DTMF signals to a telephone on an fxs module in the Adtran. Asterisk is seeing the call answered when the T-1 is picked up by the Adtran not when the ringing phone is answered. This means the digits have already been sent by the time the ringing phone is answered. Does anyone have an idea on how to signal this correctly? SendDTMF maybe? http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
Well, setting up queues for every user is one option, but it's troublesome. Also, nearly all commercial VoIP origination platforms I've seen, including that of a former Vonage-like employer, support concurrent contacts in their registrar. I guess to really do this as a matter of implementational fact, one would have to either modify the Asterisk source somewhat extensively, or use a separate service to actually hold the contacts that does allow concurrent registrants, such as OpenSER. Sort of like a homespun session border controller. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
On Wed, 5 Dec 2007, Trevor G. Hammonds wrote: As of 19th September 2007, ATT discontinued the service due to the unavailability of parts for the 1960s-era Audichron equipment, and declining use of the service. I don't believe for a minute that it was discontinued due to lack of parts. I think anybody on this list could whack out an Asterisk box to replace it :) I think the market value of the xx,xxx DNISs versus a free service is a much more likely motivation. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
El Mie, 5 de Diciembre de 2007, 11:45, Michael Melia Jr. escribió: Does anyone know how I could integrate my Asterisk setup with Outlook so that when I click on a phone number is my outlook address book it will dial the number and ring my SIP phone so that I can just pick it up? I am interested in this integration for WinXP with Outlook 2003 and WInVista with Outlook 2007. Try OutCall: http://outcall.sourceforge.net/ Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
I'd look at a program called Outcall. I believe this will handle everything you'll need. Michael Melia Jr. wrote: Does anyone know how I could integrate my Asterisk setup with Outlook so that when I click on a phone number is my outlook address book it will dial the number and ring my SIP phone so that I can just pick it up? I am interested in this integration for WinXP with Outlook 2003 and WInVista with Outlook 2007. Thanks, Michael ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pstn call waiting and zap
Patricio Valarezo Lozano wrote: Hi, I hope someone could help me, i have a x100p interface for testing purpose and on each incomming call I redirect the call to handytone 388 atas, the problem comes when i'm during a call and another call comes in, i hear the call waiting beep (comming from the zap channel), but I can't catch the call as usually using flash+2 (my pstn call wait sequence), because when i flash the sip channel i get the tone for transfering. How should i get the call ? i was trying to flash the zap channel using zapflash but it did not work. thanks a lot for your time, i hope have exposed the problem crearly. PV Have you checked into the console output when you try this? Please paste it to us. Thanks! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
Sam, Thank you for the suggestion. That is pretty much what I ended up doing for myself anyway; the real issue is standardising it and doing it on a mass scale for all users of a platform. -- Alex On Wed, 5 Dec 2007, Lutgring, Sam wrote: Alex; I would suggest simply registering them as separate or unique phones and then ringing multiple phones from the same extension using the . This way both phones will ring and you can answer based on which one is local to you. I do this with my desk phone and my X-lite soft phone. Here is what it looks like: SIP.CONF [sam-X-1433]; This is my X-lite phone type=friend username=sam-X-1433 -SNIP- [sam-G-1433]; This is my desk phone type=friend username=sam-G-1433 -SNIP- EXTENSIONS.CONF exten = 1433,1,Dial(SIP/sam-G-1433SIP/sam-X-1433,22,Tt) Hope you find this to be useful. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Wednesday, December 05, 2007 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple contacts. Well, setting up queues for every user is one option, but it's troublesome. Also, nearly all commercial VoIP origination platforms I've seen, including that of a former Vonage-like employer, support concurrent contacts in their registrar. I guess to really do this as a matter of implementational fact, one would have to either modify the Asterisk source somewhat extensively, or use a separate service to actually hold the contacts that does allow concurrent registrants, such as OpenSER. Sort of like a homespun session border controller. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
On Wed, Dec 05, 2007 at 11:07:01AM -0500, SIP wrote: IM is one of those few scenarios where I think that I'd NOT want to have possibly multiple logins at the same time. The last thing I need is to have one half of a conversation on a random machine that I forgot to log out of -- if nothing else, just for the space it takes up. XMPP (Jabber) actually works with mutiple clients connected to the same address. But they have to explicitly create themselves separate resources on the local server. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
On Wed, Dec 05, 2007 at 12:26:46PM -0500, Jared Smith wrote: On Wed, 2007-12-05 at 11:45 -0500, Michael Melia Jr. wrote: Does anyone know how I could integrate my Asterisk setup with Outlook One of the more popular ones seems to be Outcall, which is now open-source and available from http://outcall.sourceforge.net. I haven't tried it personally, so your mileage may vary. According to its documentation, outcall works by granting each user practically full control over Asterisk through the manager interface, right? Or at least the call write permission, which is pretty close to full control. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
Tony Mountifield wrote: In article [EMAIL PROTECTED], Ryan Burke [EMAIL PROTECTED] wrote: I just was looking over the app_waitutil.c and am confused you add 500 to tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec + 500) / 1000);? It's just doing a standard round to nearest integer division, by adding half the divisor to the dividend before dividing. Without that, you just get round down instead. Cheers Tony That's right. ast_safe_sleep() has a resolution of msec, but gettimeofday() returns the time in usec, so conversion to the nearest whole msec is necessary. -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No timezone in Voicemail email?
Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY 14624 Office: 888-865-0065 Ext. 202 Mobile: (585) 721-8679 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and TDM400P
Hi, I have a problem with a TDM400P card configuration. Incoming calls are answered by asterisk, asterisk place the call on the destination ATA/analog-phone, the phone begins to ring and when our recepcionist pickup the phone to play a welcome message, she nothing hear on the line during five or six seconds repeating the message three or four times until someone appear on the phone. I was playing with Wait application but nothing changes this issue. On the other hand I disabled callerid from zapata.conf but this problem continues. I'm using Asterisk 1.2.24 on a debian system I hope have exposed the problem crearly. Thanks for any idea to solve this issue. Cheers Alejandro González ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
I would recommend Activa TSP as I prefer its Outook integration than Outcall's one : - you're not limited to local contact folders, - it doesn't need to import contacts - GUI is simple. It's based on TAPI and AMI. A bug in AstManProxy prevent it to be used with it. When you pick a Contact in Outlook, you select Actions|Call this contact... and your own hardphone starts to ring. I got several bugs (from Microsoft, I would say) configuring it on XP : it disappeared with a fresh new XP install. Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softswitch digim
Bill Hackensack wrote: On Dec 2, 2007 3:42 PM, Carlos Rojas [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm looking the softswitch in digium website, anyone test the softswitch? Nope. No one has tested it or used it. Try the one at cisco.com http://cisco.com. Digium has a product they call 'asterisk' that might work out well for you. I think it's a softswitch :) You might read a little more about that one. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
Steve Totaro wrote: SIP wrote: Steve Totaro wrote: Alex Balashov wrote: I'm sure this has been asked a million times before, but is there an easy wa to have Asterisk register more than one (distinct) contact binding concurrently? The goal is to have two phones register with the same credentials from different locations and consistently and reliably ring on inbound calls, irrespective of their registration intervals and so on. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 A million and one now, check the archives. No you cannot and why would you want to? The device that registers last will ring. Just set the phones up in a ring group or even a ring all queue. Thanks, Steve Totaro ___ Why would you WANT to? Well heck, I can think of a lot of reasons -- not the least of which being able to freely log in from anywhere at anytime with multiple phones (the wifi sip phone from the coffee shop, the desk phone at the office, the phone at home, the new phone I just picked up at lunchtime) without having to configure a device entry for each and every one of them and modify extensions.conf to ring multiple devices for each and every phone I add or remove from the list. In short, flexibility. The problem with this question is the way Asterisk thinks of phones to the way many people think of logins. To Asterisk, phones are devices -- separate entities for which there should be an entry each time. To those of us NOT migrating into Asterisk from the traditional PBX world, this is somewhat of a foreign concept. The idea that everywhere we log in from must be a unique device that has to be configured to be allowed to log in is somewhat weird in a world of mobility. In the days of terminals all connecting to a central hub, it made more sense. But in the days of internet cafes, library computers, wi-fi everywhere, etc., it's just not a compatible concept. Who wants to reconfigure his VoIP box every time he goes to a new computer with a new softphone, for instance? So while it may make absolutely PERFECT sense in the realm of Asterisk, as Asterisk is a PBX system and that's how PBX systems think, I'm always surprised at the number of people who simply don't understand why people ask this question. A lot. :) N. Every machine in a in a Windows environment must be configured to join a domain. A user must also be setup in that domain to log in. It is more secure to lock that user into a single login session so that if they are logged in at one machine, they cannot login somewhere else. Think of it like that. Flexibility is not always best practice nor secure. I do not see how internet cafes and wifi have anything to to do with anything. If you go to any of these places with your softphone or wifi phone, they should work. I am not sure how you would expect a computer to just know how to configure itself other than setting up a download site with a provisioning tool. AFAIK, computers cannot read minds yet, nor just configure themselves without human intervention. If you want to be that flexible you can just configure Asterisk to allow you to auto register and use authenticate on dialing or to be really flexible, just leave it wide open until you file to file bankruptcy due to toll fraud. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Don't be melodramatic, Steve. Look at most Internet services. I can log into email from just about anywhere with any client. I don't have to set it up before hand. I can log into my workstations from any SSH client I choose (as long as I'm in an allowed network). I don't have to preconfigure which ones are allowed and set them up before hand. I can log into a web site with any browser I choose -- the web site owners, apart from a few modifications that might need to be made for formatting, don't need to configure their site for each and every browser. With SER/OpenSER, I can create a system where multiple phones can log in using the same credentials because it doesn't even CARE about the devices themselves -- just the users logging in (on our service, I have my home phone, mobile, and work phone all logged in with the same number -- it catches me anywhere I happen to be, and I don't have to make modifications to the server and reload configs every time I want to add a phone into the mix). And yet, none of this increases the fraud possibilities. It's simply the flexibility that's expected in this day and age. As long as you authenticate
Re: [asterisk-users] No timezone in Voicemail email?
Jason Martin wrote: Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. This is already fixed in 1.4.15. -- Jason Parker Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
On Wed, 2007-12-05 at 11:45 -0500, Michael Melia Jr. wrote: Does anyone know how I could integrate my Asterisk setup with Outlook One of the more popular ones seems to be Outcall, which is now open-source and available from http://outcall.sourceforge.net. I haven't tried it personally, so your mileage may vary. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No timezone in Voicemail email?
On Wednesday 05 December 2007 01:25:19 pm Jason Martin wrote: Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. ignore my previous message. it works properly in 1.4.15, but is broken in 1.4.14. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Popcorn ( was Re: New feature: calling all bug marshals )
Philip Prindeville wrote: Steve Edwards wrote: On Tue, 4 Dec 2007, Philip Prindeville wrote: I wanted to write a popcorn app for myself, both to learn how to script in Just out of curiosity, what does this have to do with popcorn? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 You used to be able to dial popcorn (767-2676) in any area code (at least prior to 1982) and get the current time. -Philip Not really ANY area code. That was mostly a Western US thing. In the Mid Atlantic US, CP, later Bell Atlantic, later VeriZon, it was TI-4-2525. There was NO standard throughout the Bell System, and often not even offered by independents, though some time and weather along with a short commercial were sponsored by banks and such, often recorded on an Audichron system. John Novack -- Dog is my co-pilot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SIP Microsoft Outlook Integration
Does anyone know how I could integrate my Asterisk setup with Outlook so that when I click on a phone number is my outlook address book it will dial the number and ring my SIP phone so that I can just pick it up? I am interested in this integration for WinXP with Outlook 2003 and WInVista with Outlook 2007. Thanks, Michael ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
Alex; I would suggest simply registering them as separate or unique phones and then ringing multiple phones from the same extension using the . This way both phones will ring and you can answer based on which one is local to you. I do this with my desk phone and my X-lite soft phone. Here is what it looks like: SIP.CONF [sam-X-1433]; This is my X-lite phone type=friend username=sam-X-1433 -SNIP- [sam-G-1433]; This is my desk phone type=friend username=sam-G-1433 -SNIP- EXTENSIONS.CONF exten = 1433,1,Dial(SIP/sam-G-1433SIP/sam-X-1433,22,Tt) Hope you find this to be useful. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Wednesday, December 05, 2007 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple contacts. Well, setting up queues for every user is one option, but it's troublesome. Also, nearly all commercial VoIP origination platforms I've seen, including that of a former Vonage-like employer, support concurrent contacts in their registrar. I guess to really do this as a matter of implementational fact, one would have to either modify the Asterisk source somewhat extensively, or use a separate service to actually hold the contacts that does allow concurrent registrants, such as OpenSER. Sort of like a homespun session border controller. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
Ira wrote: At 11:58 PM 12/4/2007, you wrote: You used to be able to dial popcorn (767-2676) in any area code (at least prior to 1982) and get the current time. I thought it was UL3-2121 when I was younger and occasionally if that was the only number in the UL3 prefix, dialing just UL3 was enough to get the time. Ira Who would have suspected that I'd be opening such a floodgate of nostalgia? :-) Anyway, can anyone tell me what other steps I might need to take to get my feature considered for future integration? Thanks, -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
At 11:58 PM 12/4/2007, you wrote: You used to be able to dial popcorn (767-2676) in any area code (at least prior to 1982) and get the current time. I thought it was UL3-2121 when I was younger and occasionally if that was the only number in the UL3 prefix, dialing just UL3 was enough to get the time. Ira ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk server and DSCP QOS
We're using 184 here (aka TOS 5/EF). Not set by iptables though, instead it is set in sip.conf (tos_sip/tos_audio) and on our Polycom/Cisco phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Johnson Sent: Wednesday, December 05, 2007 12:49 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk server and DSCP QOS Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using the Linksys SGE2000P POE switch which supports QOS via DSCP. Thanks a lot for any info. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Text-To-Speech synthesizer--help required
At 03:13 12/5/2007, srinivas Antarvedi wrote: Hello users, Actually i wanted to implement Text-To-Speech engine from cepstral voice using swift application i tried the documentation of doing this and i was unsuccessful at doing this work with asterisk can anybody please help me out finding the solution to installation thanks in advacnce srinivas Antarvedi Looking for this? http://www.voip-info.org/wiki-Asterisk+cmd+Festival http://www.voip-info.org/wiki-Asterisk+Festival+installation ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk server and DSCP QOS
Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using the Linksys SGE2000P POE switch which supports QOS via DSCP. Thanks a lot for any info. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729/MOH Quality
Yes, it is in queues but there isn't anywhere to move them :) Instead we went ahead and generated whitenoise files just above the silence supression threshold to use as an alternate which is a little easier on the ears. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, November 30, 2007 16:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729/MOH Quality If the majority of the MoH is queues, move the location of the queue. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk server and DSCP QOS
Thanks, Darryl, To clarify: in /etc/asterisk/sip.conf you have the lines: tos_sip=cs3; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. and in your Polycom configuration [I'm using Polycom's sip 2.2.0] you have something like (this is the one I'm uncertain about): QOS Ethernet RTP qos.ethernet.rtp.user_priority=5/ CallControl qos.ethernet.callControl.user_priority=5/ Other qos.ethernet.other.user_priority=2/ /Ethernet IP RTP qos.ip.rtp.dscp=184 qos.ip.rtp.min_delay=1 qos.ip.rtp.max_throughput=1 qos.ip.rtp.max_reliability=0 qos.ip.rtp.min_cost=0 qos.ip.rtp.precedence=5/ CallControl qos.ip.callControl.dscp=184 qos.ip.callControl.min_delay=1 qos.ip.callControl.max_throughput=0 qos.ip.callControl.max_reliability=0 qos.ip.callControl.min_cost=0 qos.ip.callControl.precedence=5/ /IP /QOS Thanks again! Steve Darryl Duncan wrote: We're using 184 here (aka TOS 5/EF). Not set by iptables though, instead it is set in sip.conf (tos_sip/tos_audio) and on our Polycom/Cisco phones. -Original Message- Subject: [asterisk-users] Asterisk server and DSCP QOS Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using the Linksys SGE2000P POE switch which supports QOS via DSCP. Thanks a lot for any info. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No timezone in Voicemail email?
On Wednesday 05 December 2007 01:25:19 pm Jason Martin wrote: Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. i just noticed this as well. using 1.4.15 from atrpms.net. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7960 to 2 SIP servers?
Is it possible for a Cisco 7960 phone with SIP firmware to connect to 2 different SIP servers @ the same time? I currently have an asterisk box @ home with several sip extensions and a Nortel C15k phoneswitch at work (not the pbx, the full phone switch). I can connect from the SIP phone to the Nortel phone switch, but cannot make asterisk talk to it at all (if anyone has any ideas on this one, I'd be hugely grateful). So I thought if I could have the cisco ip phone on my desk talk to both servers (like a line1 is my home asterisk server, line 2 is the nortel switch) I'd be all set. Does anyone know if this is possible, and if so how to do it? Thanks in advance Shawn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk server and DSCP QOS
Looks fine to me, you only need to specify DSCP or TOS (may need to check the manual for which it takes first). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Johnson Sent: Wednesday, December 05, 2007 14:02 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisk server and DSCP QOS Thanks, Darryl, To clarify: in /etc/asterisk/sip.conf you have the lines: tos_sip=cs3; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. and in your Polycom configuration [I'm using Polycom's sip 2.2.0] you have something like (this is the one I'm uncertain about): QOS Ethernet RTP qos.ethernet.rtp.user_priority=5/ CallControl qos.ethernet.callControl.user_priority=5/ Other qos.ethernet.other.user_priority=2/ /Ethernet IP RTP qos.ip.rtp.dscp=184 qos.ip.rtp.min_delay=1 qos.ip.rtp.max_throughput=1 qos.ip.rtp.max_reliability=0 qos.ip.rtp.min_cost=0 qos.ip.rtp.precedence=5/ CallControl qos.ip.callControl.dscp=184 qos.ip.callControl.min_delay=1 qos.ip.callControl.max_throughput=0 qos.ip.callControl.max_reliability=0 qos.ip.callControl.min_cost=0 qos.ip.callControl.precedence=5/ /IP /QOS Thanks again! Steve Darryl Duncan wrote: We're using 184 here (aka TOS 5/EF). Not set by iptables though, instead it is set in sip.conf (tos_sip/tos_audio) and on our Polycom/Cisco phones. -Original Message- Subject: [asterisk-users] Asterisk server and DSCP QOS Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using the Linksys SGE2000P POE switch which supports QOS via DSCP. Thanks a lot for any info. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
Thanks for the suggestions so far. I don't like the idea that I have to give full control with OutCall but it seems to be the case with most of the solutions out there. I have downloaded and tested OutCall on Windows Vista and Outlook 2007. It doesn't seem to work 100% with Outlook 2007. Program looks promising but needs some revision for latest version of Outlook. It seems a lot of the issues I saw have been posted to OutCalls forum and they are working on the Outlook 2007 with Exchange integration. Maybe 1.5 will be what I am looking for. I am going to test on XP and Outlook 2003 later. I have not tried Activa TSP yet. Anyone have any feedback on that? Thanks, Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, December 05, 2007 12:55 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration On Wed, Dec 05, 2007 at 12:26:46PM -0500, Jared Smith wrote: On Wed, 2007-12-05 at 11:45 -0500, Michael Melia Jr. wrote: Does anyone know how I could integrate my Asterisk setup with Outlook One of the more popular ones seems to be Outcall, which is now open-source and available from http://outcall.sourceforge.net. I haven't tried it personally, so your mileage may vary. According to its documentation, outcall works by granting each user practically full control over Asterisk through the manager interface, right? Or at least the call write permission, which is pretty close to full control. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pstn call waiting and zap
Mojo with Horan Company, LLC wrote: Patricio Valarezo Lozano wrote: Hi, I hope someone could help me, i have a x100p interface for testing purpose and on each incomming call I redirect the call to handytone 388 atas, the problem comes when i'm during a call and another call comes in, i hear the call waiting beep (comming from the zap channel), but I can't catch the call as usually using flash+2 (my pstn call wait sequence), because when i flash the sip channel i get the tone for transfering. How should i get the call ? i was trying to flash the zap channel using zapflash but it did not work. thanks a lot for your time, i hope have exposed the problem crearly. PV Have you checked into the console output when you try this? Please paste it to us. Thank you, i was digging for a solution and i have use application map in features.conf, as follow: zapflash = 22,callee,flash,() so wen i try to flash the channel i get this output on the console: # pressing 22 during a call -- Feature Found: zapflash exten: zapflash -- Flashed channel Zap/1-1 # using only flash button -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Stopped music on hold on Zap/1-1 but it did'nt work either, i thing i'm close to the solution, my PSTN callwaiting sequence is flash+2... guessing thanks for your help PatoVala -- patoVala Linux User#280504 Hablando en http://www.elprimoalcahuete.com Knghtbrd the problem with the GNU coding standards is they ASSUME that everyone in the world uses emacs.. If that were the case, free software would die because we would all have wrist problems like RMS by now and no longer be able to code. ; ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
SIP wrote: Every machine in a in a Windows environment must be configured to join a domain. A user must also be setup in that domain to log in. It is more secure to lock that user into a single login session so that if they are logged in at one machine, they cannot login somewhere else. Think of it like that. Flexibility is not always best practice nor secure. I do not see how internet cafes and wifi have anything to to do with anything. If you go to any of these places with your softphone or wifi phone, they should work. I am not sure how you would expect a computer to just know how to configure itself other than setting up a download site with a provisioning tool. AFAIK, computers cannot read minds yet, nor just configure themselves without human intervention. If you want to be that flexible you can just configure Asterisk to allow you to auto register and use authenticate on dialing or to be really flexible, just leave it wide open until you file to file bankruptcy due to toll fraud. Thanks, Steve Totaro Don't be melodramatic, Steve. Look at most Internet services. I can log into email from just about anywhere with any client. I don't have to set it up before hand. I can log into my workstations from any SSH client I choose (as long as I'm in an allowed network). I don't have to preconfigure which ones are allowed and set them up before hand. I can log into a web site with any browser I choose -- the web site owners, apart from a few modifications that might need to be made for formatting, don't need to configure their site for each and every browser. With SER/OpenSER, I can create a system where multiple phones can log in using the same credentials because it doesn't even CARE about the devices themselves -- just the users logging in (on our service, I have my home phone, mobile, and work phone all logged in with the same number -- it catches me anywhere I happen to be, and I don't have to make modifications to the server and reload configs every time I want to add a phone into the mix). And yet, none of this increases the fraud possibilities. It's simply the flexibility that's expected in this day and age. As long as you authenticate SOMEhow, you're authenticated. That's kind of the idea behind authentication. If username/password authentication isn't enough, then perhaps there's a flaw in your auth process. It's not an unreasonable question to ask why you have to authenticate BOTH the device AND the user using the device when you could just say devices are allowed to log in as long as the user is and allow any and all of them if you so CHOOSE. You might choose not to. But it's not unreasonable to want that choice. IM is one of those few scenarios where I think that I'd NOT want to have possibly multiple logins at the same time. The last thing I need is to have one half of a conversation on a random machine that I forgot to log out of -- if nothing else, just for the space it takes up. However, with phones? One can be reasonably certain that I'm in control of the phones I'm logging in from. If I'm not, then the administrator should choose to disallow multiple logins from the same ID. However, if so, where's the harm in allowing it? I just don't get the whole FUD issue with this. I understand that it's simply part of the way PBX systems work... but discounting the option as 'dangerous' is just masking the issue. N. Not sure what the whole FUD thing is but you do seem very passionate about it Short answer, no it cannot be done, don't like it? Use SER as you say or change the Asterisk code, it is opensource after all. It is what it is. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
In article [EMAIL PROTECTED], Ryan Burke [EMAIL PROTECTED] wrote: I just was looking over the app_waitutil.c and am confused you add 500 to tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec + 500) / 1000);? It's just doing a standard round to nearest integer division, by adding half the divisor to the dividend before dividing. Without that, you just get round down instead. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use of slin as a codec
Partially answering my own question, it looks like slin is a 128 kbps codec. Peter From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Whisker, Peter Sent: 05 December 2007 16:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Use of slin as a codec Where bandwidth is not an issue but good call quality is, is there any theoretical quality improvement to be had by using slin as the codec over an inter-Asterisk IAX trunk rather than a-law (or u-law in the US). Does anyone know what the slin bandwidth is (compared to 64 kbps a-law). Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
Hi.
I wanted to write a popcorn app for myself, both to learn how to
script in extensions.conf, but also because it was something handy.
Along the way, I found myself doing something like:
[popcorn]
exten = s,1,Set(FUTURETIME=$[${EPOCH} + 10])
...
exten = s,n,While(${EPOCH} ${FUTURETIME})
exten = s,n,Wait(0.01)
exten = s,n,EndWhile()
exten = s,n,Play(beep)
exten = s,n,Hangup()
and hating myself for it (my Asterisk runs on a 500MHz Geode LX).
So I decided it would be useful (in general, and educational for me in
particular) to write a WaitUntil() application instead.
Well, I've done that.
I was going to file a bug and then post a fix to get their feature in,
but the Bug guidelines seem to be pretty clear that this is not the way
to go.
So, I'm posting here instead.
The example to paste into the documentation or extensions.conf is:
[popcorn]
exten = s,1,Answer()
; the amount of delay is set for English; you may need to adjust this time
; for other languages is there's no pause before the synchronizing beep.
exten = s,n,Set(FUTURETIME=$[${EPOCH} + 11])
exten = s,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
exten = s,n,SayPhonetic(z)
exten = s,n,SayUnixTime(${FUTURETIME},,HNS)
exten = s,n,Playback(local)
exten = s,n,WaitUntil(${FUTURETIME})
exten = s,n,Playback(beep)
exten = s,n,Return()
I invoke it as:
exten = 712,1,Gosub(popcorn,s,1)
exten = 712,n,Hangup()
And lastly, attached is the source for app_waituntil.c.
It's fairly straightforward, and not very big.
But hopefully useful.
Oh, before I forget: it does require the recording of one additional
phrase,
either local or localtime. I've used local in my example above.
And
I read out the time first as GMT/UT (because I travel a lot), and then in
the
timezone of my PBX...
-Philip
Philip,
I just was looking over the app_waitutil.c and am confused you add 500 to
tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec
+ 500) / 1000);?
Ryan
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI error
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Good Morning, My problem was that the context wasn't the same in my voicemail.conf and in my sip.conf!! One was 'default' and the other 'device' I have put 'default' everywhere and it's working! Have a nice day Jared Smith a écrit : On Tue, 2007-12-04 at 17:20 +0100, Marc LEURENT wrote: It's just that I received SIP notify message saying that there is nothing in the voicemail even when there is a message... Do you have a mailbox defined for the SIP device in sip.conf? If you don't, Asterisk has no way of matching up a mailbox to a particular SIP device. -Jared Smith ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (Darwin) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHVmcvN4+o+2LtdFwRAkdSAJ9KPkr9NGc9nm+wIFGUofcE4nxQnACfRJeL HakgTsDpHM7QCCyvzPI0440= =J5cK -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange ISDN-problem with incoming calls out of the same city
Does this number (you are dialing) has been ported from a different Telco? When you dial from the other city and you get service not available you may be dialing from a different Telco that either has no route aggreement for the dialed network, or the number portability database (of Out of city Operator) is not up to date. Can you confirm this two things? On Dec 5, 2007 10:05 PM, Stefan Guenther [EMAIL PROTECTED] wrote: Hi, after I fixed my problem with the playback() application, I now have the next strange one. When I dial the number of our client, located in another town, I get a connection to the asterisk server, I can talk to my client or listen to his mailbox. If some in the town of this client calls him, he gets the ISDN error service not available. Out office is connected to he office via vpn, and so I connected a sip phone to his asterisk server. Now this phone is in the same town and when I dial his external number (to make sure it is an ISDN connection), I hear a single ring tone, then the phone is connected but I here nothing. Here is the output of capi debug: -- Executing [EMAIL PROTECTED]:1] Set(SIP/user1-0827eb08, CALLERID(num)=7253940397) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/user1-0827eb08, CAPI/g1/7253940397:940388|60|tr) in new stack -- Called g1/7253940397:940388 == ISDN5#01: Incoming call '072539403' - '940388' -- Executing [EMAIL PROTECTED]:1] Answer(CAPI/ISDN5#01/940388-3, ) in new stack == ISDN5#01: Answering for 940388 -- Executing [EMAIL PROTECTED]:2] Wait(CAPI/ISDN5#01/940388-3, 1) in new stack == ISDN5#01: Setting up echo canceller (PLCI=0x305, function=1, options=4, tail=64) == ISDN5#01: Setting up DTMF detector (PLCI=0x305, flag=1) -- ISDN5#01: Echo canceller successfully set up (PLCI=0x305) == ISDN5#02: Setting up echo canceller (PLCI=0x205, function=1, options=4, tail=64) == ISDN5#02: Setting up DTMF detector (PLCI=0x205, flag=1) -- CAPI/ISDN5#02/940388-2 answered SIP/user1-0827eb08 -- ISDN5#02: Echo canceller successfully set up (PLCI=0x205) -- Executing [EMAIL PROTECTED]:3] GotoIfTime(CAPI/ISDN5#01/940388-3, 17:00-18:00|*|*|*?from-extern|940388|6) in new stack -- Executing [EMAIL PROTECTED]:4] Dial(CAPI/ISDN5#01/940388-3, SIP/VERKAUF|20|tr) in new stack -- Called VERKAUF -- SIP/VERKAUF-082841a0 is ringing -- SIP/VERKAUF-082841a0 is ringing -- SIP/VERKAUF-082841a0 is ringing == ISDN5#02: CAPI Hangingup for PLCI=0x205 in state 2 == Spawn extension (local, 940388, 2) exited non-zero on 'SIP/user1-0827eb08' What is the difference between an isdn call starting in the same area and a call from my office? We are using a EICON DIVA Server 4 BRI with the current driver and an asterisk 1.4.13. Here is the start of the capi.conf [General] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de immediate=yes faxdetect=off [ISDN1] incomingmsn=* context=from-extern ntmode=yes controller=1 group=1 callgroup=1 accountcode=ISDN1 echocancel=yes echosquelch=1 echotail=64 devices=2 Thanks for any suggestions, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
On Wed, 5 Dec 2007, SIP wrote: I just don't get the whole FUD issue with this. I understand that it's simply part of the way PBX systems work... but discounting the option as 'dangerous' is just masking the issue. I would tend to agree. One of the key value propositions proffered by VoIP in terms of technological transformation is a movement to media-agnostic convergent networking that includes all sorts of presence and find-me-follow-me functionality as a basic element of the reachability methodology. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP-Realtime and sip reload
I use SIP-Realtime to store my SIP-users and I keep the informations about the SIP-Providers my Asterisk registers to in sip.conf. I'm running into the following problem. If I set rtcachefriends=yes because I want to use MWI and run a sip reload because I changed something in sip.conf, Asterisk forgets about all registrations of the users which are all unavailable after that. How can I use rtcachefriends=yes to allow MWI (isn't it needed for NAT-keepalive as well?) and don't break everything with a sip reload? The short answer is, this is how it works, don't reload sip.conf or loose your cache. You can set your phone registration time lower that 3600 so phones re-register quicker. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
Steve Edwards wrote: On Wed, 5 Dec 2007, Trevor G. Hammonds wrote: As of 19th September 2007, ATT discontinued the service due to the unavailability of parts for the 1960s-era Audichron equipment, and declining use of the service. I don't believe for a minute that it was discontinued due to lack of parts. I think anybody on this list could whack out an Asterisk box to replace it :) I think the market value of the xx,xxx DNISs versus a free service is a much more likely motivation.-- Indeed, in fact the successor to Audichron still manufactures equipment and supports it. This a pure and simple issue of corporate greed on the part of SBC/att. John Novack Dog is my co-pilot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disturbance noise in the background for digium card
bilal ghayyad wrote: 1) Sometimes when I call to the PBX, I hear like modem sound and after little it disapear. 2) There is a disturbance in the background (like the channel radio disturbance that might happen if the frequency was not captured well), and that disturbance appear much more when Asterisk goes via IP Trunk. Is it a configuration issue or it might be a digium card defection so need to be replaced? It is very likely that your issues can be solved by adjusting configuration. Please contact [EMAIL PROTECTED] for help with what needs changing. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] redirected call failure
Hi, I have the following setup: (sip clients) -internet- asterisk A -IAX- asterisk B -PRI- (pstn) This works fine for regular calls sip-pstn. the calls go through perfectly. However, when one of the sip clients (a snom320) is set to redirect to the pstn, then all I hear is congestion tones when I call that sip client. I am at a loss to see why the redirected call would fail, when the redirecting sip client can dial the same number successfully. In the pri debug trace below (from asterisk host B), the sip phone at extension 340340 (local ext) is set to redirect to 131166 (a pstn number). I notice that 340340 appears in the PRI debug on asterisk host B even though it has no direct contact with the sip client. The callerid is set exactly the same as a regular call from ext 340340 would be. What else is different from asterisk host B's point of view in this case? -- Accepting AUTHENTICATED call from 192.168.10.101: requested format = g729, requested prefs = (g729), actual format = g729, host prefs = (g729), priority = caller -- Executing [EMAIL PROTECTED]:4] Goto(IAX2/holly-v2-g729-3, terminate-pri|131166|1) in new stack -- Goto (terminate-pri,131166,1) -- Executing [EMAIL PROTECTED]:3] Dial(IAX2/holly-v2-g729-3, ZAP/g1/131166|240) in new stack -- Making new call for cr 44126 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=50 Call Ref: len= 2 (reference 11358/0x2C5E) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 0c 00 83 30 33 39 30 31 33 31 37 30 30] Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation allowed of network provided number (3) '0390131700' ] [70 07 80 31 33 31 31 36 36] Called Number (len= 9) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '131166' ] [74 09 00 01 8f 33 34 30 33 34 30] Redirecting Number (len=11) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Ext: 0 Presentation: Presentation permitted, user number passed network screening (1) Ext: 1 Reason: Forwarded unconditionally (15) '340340' ] [a1] Sending Complete (len= 1) q931.c:2881 q931_setup: call 44126 on channel 1 enters state 1 (Call Initiated) -- Called g1/131166 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 11358/0x2C5E) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Iden Can anyone suggest what might be causing this behaviour? Thanks, Mostyn. -- [EMAIL PROTECTED] * ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange ISDN-problem with incoming calls out of the same city
Hi, after I fixed my problem with the playback() application, I now have the next strange one. When I dial the number of our client, located in another town, I get a connection to the asterisk server, I can talk to my client or listen to his mailbox. If some in the town of this client calls him, he gets the ISDN error service not available. Out office is connected to he office via vpn, and so I connected a sip phone to his asterisk server. Now this phone is in the same town and when I dial his external number (to make sure it is an ISDN connection), I hear a single ring tone, then the phone is connected but I here nothing. Here is the output of capi debug: -- Executing [EMAIL PROTECTED]:1] Set(SIP/user1-0827eb08, CALLERID(num)=7253940397) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/user1-0827eb08, CAPI/g1/7253940397:940388|60|tr) in new stack -- Called g1/7253940397:940388 == ISDN5#01: Incoming call '072539403' - '940388' -- Executing [EMAIL PROTECTED]:1] Answer(CAPI/ISDN5#01/940388-3, ) in new stack == ISDN5#01: Answering for 940388 -- Executing [EMAIL PROTECTED]:2] Wait(CAPI/ISDN5#01/940388-3, 1) in new stack == ISDN5#01: Setting up echo canceller (PLCI=0x305, function=1, options=4, tail=64) == ISDN5#01: Setting up DTMF detector (PLCI=0x305, flag=1) -- ISDN5#01: Echo canceller successfully set up (PLCI=0x305) == ISDN5#02: Setting up echo canceller (PLCI=0x205, function=1, options=4, tail=64) == ISDN5#02: Setting up DTMF detector (PLCI=0x205, flag=1) -- CAPI/ISDN5#02/940388-2 answered SIP/user1-0827eb08 -- ISDN5#02: Echo canceller successfully set up (PLCI=0x205) -- Executing [EMAIL PROTECTED]:3] GotoIfTime(CAPI/ISDN5#01/940388-3, 17:00-18:00|*|*|*?from-extern|940388|6) in new stack -- Executing [EMAIL PROTECTED]:4] Dial(CAPI/ISDN5#01/940388-3, SIP/VERKAUF|20|tr) in new stack -- Called VERKAUF -- SIP/VERKAUF-082841a0 is ringing -- SIP/VERKAUF-082841a0 is ringing -- SIP/VERKAUF-082841a0 is ringing == ISDN5#02: CAPI Hangingup for PLCI=0x205 in state 2 == Spawn extension (local, 940388, 2) exited non-zero on 'SIP/user1-0827eb08' What is the difference between an isdn call starting in the same area and a call from my office? We are using a EICON DIVA Server 4 BRI with the current driver and an asterisk 1.4.13. Here is the start of the capi.conf [General] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de immediate=yes faxdetect=off [ISDN1] incomingmsn=* context=from-extern ntmode=yes controller=1 group=1 callgroup=1 accountcode=ISDN1 echocancel=yes echosquelch=1 echotail=64 devices=2 Thanks for any suggestions, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
Ring all queues would be easier I would think. Thanks, Steve Totaro Alex Balashov wrote: Sam, Thank you for the suggestion. That is pretty much what I ended up doing for myself anyway; the real issue is standardising it and doing it on a mass scale for all users of a platform. -- Alex On Wed, 5 Dec 2007, Lutgring, Sam wrote: Alex; I would suggest simply registering them as separate or unique phones and then ringing multiple phones from the same extension using the . This way both phones will ring and you can answer based on which one is local to you. I do this with my desk phone and my X-lite soft phone. Here is what it looks like: SIP.CONF [sam-X-1433]; This is my X-lite phone type=friend username=sam-X-1433 -SNIP- [sam-G-1433]; This is my desk phone type=friend username=sam-G-1433 -SNIP- EXTENSIONS.CONF exten = 1433,1,Dial(SIP/sam-G-1433SIP/sam-X-1433,22,Tt) Hope you find this to be useful. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Wednesday, December 05, 2007 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multiple contacts. Well, setting up queues for every user is one option, but it's troublesome. Also, nearly all commercial VoIP origination platforms I've seen, including that of a former Vonage-like employer, support concurrent contacts in their registrar. I guess to really do this as a matter of implementational fact, one would have to either modify the Asterisk source somewhat extensively, or use a separate service to actually hold the contacts that does allow concurrent registrants, such as OpenSER. Sort of like a homespun session border controller. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New feature: calling all bug marshals
In article [EMAIL PROTECTED], Ryan Burke [EMAIL PROTECTED] wrote: I just was looking over the app_waitutil.c and am confused you add 500 to tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec + 500) / 1000);? It's just doing a standard round to nearest integer division, by adding half the divisor to the dividend before dividing. Without that, you just get round down instead. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org I see, sorry it was a brain fart... Thanks! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
Steve Totaro wrote: Alex Balashov wrote: I'm sure this has been asked a million times before, but is there an easy wa to have Asterisk register more than one (distinct) contact binding concurrently? The goal is to have two phones register with the same credentials from different locations and consistently and reliably ring on inbound calls, irrespective of their registration intervals and so on. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 A million and one now, check the archives. No you cannot and why would you want to? The device that registers last will ring. Just set the phones up in a ring group or even a ring all queue. Thanks, Steve Totaro ___ Why would you WANT to? Well heck, I can think of a lot of reasons -- not the least of which being able to freely log in from anywhere at anytime with multiple phones (the wifi sip phone from the coffee shop, the desk phone at the office, the phone at home, the new phone I just picked up at lunchtime) without having to configure a device entry for each and every one of them and modify extensions.conf to ring multiple devices for each and every phone I add or remove from the list. In short, flexibility. The problem with this question is the way Asterisk thinks of phones to the way many people think of logins. To Asterisk, phones are devices -- separate entities for which there should be an entry each time. To those of us NOT migrating into Asterisk from the traditional PBX world, this is somewhat of a foreign concept. The idea that everywhere we log in from must be a unique device that has to be configured to be allowed to log in is somewhat weird in a world of mobility. In the days of terminals all connecting to a central hub, it made more sense. But in the days of internet cafes, library computers, wi-fi everywhere, etc., it's just not a compatible concept. Who wants to reconfigure his VoIP box every time he goes to a new computer with a new softphone, for instance? So while it may make absolutely PERFECT sense in the realm of Asterisk, as Asterisk is a PBX system and that's how PBX systems think, I'm always surprised at the number of people who simply don't understand why people ask this question. A lot. :) N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] My AsteriskNo unable to registration
Hi Guillermo, I am not using Realtime.., why it seems line turned on? .. how to turn it off? BTW .. I have put type=friend into my sip.conf ..but the same problem still occurs (I am unable to register the SPA-3102 ;( Regards bie - Original Message - From: Guillermo Rodriguez [EMAIL PROTECTED] To: Undisclosed.Recipients: Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 05, 2007 6:22 PM Subject: Re: [asterisk-users] My AsteriskNo unable to registration Hi Bie, You have a problem with the postgresql conexion. You are using Realtime? [Dec 5 07:39:59] ERROR[2342] res_config_pgsql.c: Postgresql RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. [Dec 5 07:39:59] DEBUG[2342] res_config_pgsql.c: Postgresql RealTime: Cannot Connect: [Dec 5 07:39:59] WARNING[2342] res_config_pgsql.c: Postgresql RealTime: Couldn't establish connection. Check debug. [Dec 5 07:39:59] DEBUG[2342] res_config_pgsql.c: Postgresql RealTime: Cannot Connect: could not connect to server: Connection refused Is the server running on host 127.0.0.1 and accepting TCP/IP connections on port 5432? Change in your sip.conf put : type=friend failed for 'xxx.xxx.xxx.xxx' - Peer is not supposed to register Tellme something. Guillermo El Miércoles, 5 de Diciembre de 2007 01:48, Newbie escribió: Hi Guillermo, enclosed please find full log file that I got it from /var/log/asterisk please help. Thanks a lot in advance Regards Winanjaya - Original Message - From: Guillermo Rodriguez [EMAIL PROTECTED] To: Newbie [EMAIL PROTECTED] Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 04, 2007 5:59 PM Subject: Re: [asterisk-users] My AsteriskNo unable to registration Yes, the log file.. El Martes, 4 de Diciembre de 2007 12:01, Newbie escribió: Hello, could you please advise .. where can I find the trace of asterisk? do you mean log file? Thanks Regards Bie - Original Message - From: Guillermo Rodriguez [EMAIL PROTECTED] To: Newbie [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 04, 2007 5:53 PM Subject: Re: [asterisk-users] My AsteriskNo unable to registration Can you put the trace of asterisk.??' When you call to 988 Thx. Guillermo El Viernes, 30 de Noviembre de 2007 10:17, Newbie escribió: Dear The Expert, I am very new with this, I have installed AsteriskNow, X-Lite as my SoftPhone, I am using SPA-3102. I have 3 extensions, me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below) My problem is, I am unable to call 998, I thought this is registration problem, (because the Linksys screen info said Registration Failed) Could any body please help? Many thanks in advance Regards Bie below is my sip.conf allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm I also had 2 extensions (me at 250 and 998 is my SPA-3102) and my users.conf goes below: [general] fullname=New User userbase=6000 hasvoicemail=yes vmsecret=1234 hassip=yes hasiax=yes hasmanager=no callwaiting=yes threewaycalling=yes callwaitingcallerid=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callgroup=1 pickupgroup=1 host=dynamic localextenlength=0 allow_aliasextns=no allow_an_extns=no hasagent=no hasdirectory=no [250] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=Winanjaya group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=250 secret=1234 threewaycalling=yes vmsecret=1234 zapchan= registeriax=yes registersip=yes canreinvite=no nat=no dtmfmode=rfc2833 disallow=all allow=all type=peer [998] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=MyLine1 group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=999 secret=1234 threewaycalling=yes vmsecret=1234 zapchan= registeriax=yes registersip=yes canreinvite=no nat=no dtmfmode=rfc2833 disallow=all allow=all type=peer [999] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=MyPSTN group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes
[asterisk-users] Polycom Soundpoint (NO LINE)
Hi, I have just configure a Soundpoint 550 to work with Asterisk, it appear Registered to the asterisk server, and appear in asterisk console with SIP SHOW PEERS, and can receive calls, but when I try to dial, it launch a tone as if not line to dial, also can not stream audio to the other end when answer the received call. I have read the documentation but is very confuse. This phone have a Outbound Proxy (where I put the asterisk IP and port 5060) And 4 lines (where I configure the username/password and the Servers) Anyone can help me. Thanks in Advance. Atte. Ricardo Melendez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
There are probably a half dozen or more software apps that can do this. Most are free last time I checked. Google is your friend. From: Michael Melia Jr. [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 05, 2007 8:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk SIP Microsoft Outlook Integration Does anyone know how I could integrate my Asterisk setup with Outlook so that when I click on a phone number is my outlook address book it will dial the number and ring my SIP phone so that I can just pick it up? I am interested in this integration for WinXP with Outlook 2003 and WInVista with Outlook 2007. Thanks, Michael ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
Snapanumber is the best way to do this. It's a commercial app so has a license fee but works great. Cant comment about Outlook2007 but works great with 2003 for me. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Melia Jr. Sent: Wednesday, December 05, 2007 11:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk SIP Microsoft Outlook Integration Does anyone know how I could integrate my Asterisk setup with Outlook so that when I click on a phone number is my outlook address book it will dial the number and ring my SIP phone so that I can just pick it up? I am interested in this integration for WinXP with Outlook 2003 and WInVista with Outlook 2007. Thanks, Michael ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [HELP] Problems with VOIP organization
Hello! Please help me with decision problem. I need to organize voip telephony in office. I have 2 phone lines(2 physical number) for phone and fax.I need to recive call on 1 phone then redirect it to neccessary phone or fax. Can Asterisk do that ? Thanks in advance ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)
Besides grandstream-doorphone transplant surgery, no. But it does have PoE. It's cheap, especially if you already have a doorphone. If you used a GXP-2000 you can use the display and it supports XML idle screens. On Dec 4, 2007 2:53 AM, Nick Seraphin [EMAIL PROTECTED] wrote: On a similar note... has anyone ever seen a SIP-based door intercom unit? Functionality I'm looking for is... basically an outdoor rated weather resistant speaker with 1 button and microphone, when the button is pressed, it dials a specified SIP extension. Likewise, from the Asterisk box, someone can call a SIP extension and the call goes to the intercom speaker so you can initiate a conversation with the person at the door if they just rang the bell but didn't push the intercom button. Preferably something with power over ethernet support. Thanks, -- Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D - ERROR
Hi All, as good? I am trying to make a call for the Unicall channels and after the exchange of signalling and sending of the digits asterisk locks up with the sending of the signalling E and the TELCO says that asterisk would have to send signalling F, as to make for asterisk to send signalling F? The TELCO says that the signalling E is suppresor insertion of ECHO in the destination. F is end of the digits. They could help me? Best Regards Josué -- Executing Dial(SIP/1196082068-082a6b78, Unicall/g1/01197831234|90|tT) in new stack Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(1) Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Make call Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0001 - [1/ 1/Idle /Idle ] -- Called g1/01197831234 Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Dialing Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - [1/ 40/Seize /Idle ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 on - [2/ 40/Group I /Idle ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 0 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:04 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 9 on - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 9 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 7 on - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 7 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 8 on - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 8 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 3 on - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 on [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 3 off - [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1 off [2/ 40/Group I /DNIS ] Dec 5 22:48:05 WARNING[5121]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1 on - [2/ 40/Group I /DNIS ]
Re: [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)
I have seen a beta-level unit that also supported POE. With regards to non-beta hardware, standard analog doorphones work pretty well with Linksys SPA units. PaulH On Tue, 2007-12-04 at 02:53 -0500, Nick Seraphin wrote: On a similar note... has anyone ever seen a SIP-based door intercom unit? Functionality I'm looking for is... basically an outdoor rated weather resistant speaker with 1 button and microphone, when the button is pressed, it dials a specified SIP extension. Likewise, from the Asterisk box, someone can call a SIP extension and the call goes to the intercom speaker so you can initiate a conversation with the person at the door if they just rang the bell but didn't push the intercom button. Preferably something with power over ethernet support. Thanks, -- Nick On Tue, 4 Dec 2007, Doug Meredith wrote: I have searched for this without much luck. I want to be able to send public-address-like notices over VoIP phones. The LinkSys SPA-941 auto-answer support comes close to working, except that if you are currently in a call it places that call on hold without warning. I'm willing to consider a more expensive phone to solve the problem if I have to. Thanks for any help you can provide. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] astunicall-1.2.21.0.1 packages and Sangoma A104D
Hello all, as good? I am trying to use the package astunicall-1.2.21.0.1 with a Sangoma A104D card and 04 links E1 mfc/r2 in Brazil. The compilation occurred, normally and links is UP if I place in Loop and I obtain to effect called in Loop, but when I extend for the PSTN, links reports the following messages, without asterisk start calls: Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/115 - [1/ 1/Idle /Idle ] Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:2644 handle_uc_event: Unicall/115 event Far end blocked Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 - [1/ 1/Idle /Idle ] Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Detected Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Making a new call with CRN 32773 Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 1101 - [2/ 2/Idle /Idle ] Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:2644 handle_uc_event: Unicall/116 event Detected Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/115 - 1011 [1/4000/Idle /Idle ] Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 - 1011 [2/ 2/Seize ack /Seize ack ] Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:2644 handle_uc_event: Unicall/116 event Far end disconnected Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:2930 handle_uc_event: CRN 32773 - far disconnected cause=Normal, unspecified cause [31] Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Call control(6) Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Drop call(cause=Normal Clearing [16]) Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:2644 handle_uc_event: Unicall/116 event Drop call Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Call control(7) Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Release call Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 1001 - [1/1000/Clear fwd /Seize ack ] Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Release guard expired Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Destroying call with CRN 32773 Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:2644 handle_uc_event: Unicall/116 event Release call Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/116 Channel echo cancel Dec 5 22:27:13 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/115 far_unblocking_expired Dec 5 22:27:13 WARNING[3868]: chan_unicall.c:2644 handle_uc_event: Unicall/115 event Far end unblocked Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/95 - [1/ 1/Idle /Idle ] Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:2644 handle_uc_event: Unicall/95 event Far end blocked Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/96 - 0010 [1/ 1/Idle /Idle ] Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/96 Detected Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/96 Making a new call with CRN 32769 Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/96 1101 - [2/ 2/Idle /Idle ] Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:2644 handle_uc_event: Unicall/96 event Detected Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/97 - [1/ 1/Idle /Idle ] Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:2644 handle_uc_event: Unicall/97 event Far end blocked Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/98 - 1000 [1/ 1/Idle /Idle ] Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/99 - [1/ 1/Idle /Idle ] Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:2644 handle_uc_event: Unicall/99 event Far end blocked Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/100 - 1000 [1/ 1/Idle /Idle ] Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/111 - [1/ 1/Idle /Idle ] Dec 5 22:27:22 WARNING[3868]: chan_unicall.c:2644
Re: [asterisk-users] Multiple contacts.
On Dec 5, 2007 10:07 AM, SIP [EMAIL PROTECTED] wrote: Steve Totaro wrote: SIP wrote: Steve Totaro wrote: Alex Balashov wrote: I'm sure this has been asked a million times before, but is there an easy wa to have Asterisk register more than one (distinct) contact binding concurrently? The goal is to have two phones register with the same credentials from different locations and consistently and reliably ring on inbound calls, irrespective of their registration intervals and so on. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 A million and one now, check the archives. No you cannot and why would you want to? The device that registers last will ring. Just set the phones up in a ring group or even a ring all queue. Thanks, Steve Totaro ___ Why would you WANT to? Well heck, I can think of a lot of reasons -- not the least of which being able to freely log in from anywhere at anytime with multiple phones (the wifi sip phone from the coffee shop, the desk phone at the office, the phone at home, the new phone I just picked up at lunchtime) without having to configure a device entry for each and every one of them and modify extensions.conf to ring multiple devices for each and every phone I add or remove from the list. In short, flexibility. The problem with this question is the way Asterisk thinks of phones to the way many people think of logins. To Asterisk, phones are devices -- separate entities for which there should be an entry each time. To those of us NOT migrating into Asterisk from the traditional PBX world, this is somewhat of a foreign concept. The idea that everywhere we log in from must be a unique device that has to be configured to be allowed to log in is somewhat weird in a world of mobility. In the days of terminals all connecting to a central hub, it made more sense. But in the days of internet cafes, library computers, wi-fi everywhere, etc., it's just not a compatible concept. Who wants to reconfigure his VoIP box every time he goes to a new computer with a new softphone, for instance? So while it may make absolutely PERFECT sense in the realm of Asterisk, as Asterisk is a PBX system and that's how PBX systems think, I'm always surprised at the number of people who simply don't understand why people ask this question. A lot. :) N. Every machine in a in a Windows environment must be configured to join a domain. A user must also be setup in that domain to log in. It is more secure to lock that user into a single login session so that if they are logged in at one machine, they cannot login somewhere else. Think of it like that. Flexibility is not always best practice nor secure. I do not see how internet cafes and wifi have anything to to do with anything. If you go to any of these places with your softphone or wifi phone, they should work. I am not sure how you would expect a computer to just know how to configure itself other than setting up a download site with a provisioning tool. AFAIK, computers cannot read minds yet, nor just configure themselves without human intervention. If you want to be that flexible you can just configure Asterisk to allow you to auto register and use authenticate on dialing or to be really flexible, just leave it wide open until you file to file bankruptcy due to toll fraud. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Don't be melodramatic, Steve. Look at most Internet services. I can log into email from just about anywhere with any client. I don't have to set it up before hand. I can log into my workstations from any SSH client I choose (as long as I'm in an allowed network). I don't have to preconfigure which ones are allowed and set them up before hand. I can log into a web site with any browser I choose -- the web site owners, apart from a few modifications that might need to be made for formatting, don't need to configure their site for each and every browser. With SER/OpenSER, I can create a system where multiple phones can log in using the same credentials because it doesn't even CARE about the devices themselves -- just the users logging in (on our service, I have my home phone, mobile, and work phone all logged in with the same number -- it catches me anywhere I happen to be, and I don't have to make modifications to the server and reload configs every time I want to add a phone into the mix). I'm going to jump in here without reading everything... You said you can log into your
Re: [asterisk-users] [HELP] Problems with VOIP organization
Yes Asterisk can receive the calls and based either on the line the call is on or some other method route the call to a destination. That being said there are 2 things to keep in mind, the hardware cost to setup 2 incoming lines and a analog port for the fax as well as phones may be high for a 2 line setup. The other thing to keep in mind is that faxing and asterisk is one of the more complicated task. There are so many things that can break faxing that setting this up is not for the faint hearted. On Dec 5, 2007 2:49 AM, Григорий Никоноров [EMAIL PROTECTED] wrote: Hello! Please help me with decision problem. I need to organize voip telephony in office. I have 2 phone lines(2 physical number) for phone and fax.I need to recive call on 1 phone then redirect it to neccessary phone or fax. Can Asterisk do that ? Thanks in advance ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)
Quoting Paul Hales [EMAIL PROTECTED]: another option is use some sort of linux based device n770, or even an nslu2, and program a sip client to behave however you like, then just fit the thing with a usb based speaker/mic. actually, a gamepad, speaker and mic with an n770 behind a piece of glass would make a pretty damn nice doorphone / look up the person in a directory kind of application. one thing to keep in mind though - lcd screens will actually freeze and crack if they get cold enough so make sure it says on all the time or heat the case if you do something like that and its actually outside the building itself. I have seen a beta-level unit that also supported POE. With regards to non-beta hardware, standard analog doorphones work pretty well with Linksys SPA units. PaulH On Tue, 2007-12-04 at 02:53 -0500, Nick Seraphin wrote: On a similar note... has anyone ever seen a SIP-based door intercom unit? Functionality I'm looking for is... basically an outdoor rated weather resistant speaker with 1 button and microphone, when the button is pressed, it dials a specified SIP extension. Likewise, from the Asterisk box, someone can call a SIP extension and the call goes to the intercom speaker so you can initiate a conversation with the person at the door if they just rang the bell but didn't push the intercom button. Preferably something with power over ethernet support. Thanks, -- Nick On Tue, 4 Dec 2007, Doug Meredith wrote: I have searched for this without much luck. I want to be able to send public-address-like notices over VoIP phones. The LinkSys SPA-941 auto-answer support comes close to working, except that if you are currently in a call it places that call on hold without warning. I'm willing to consider a more expensive phone to solve the problem if I have to. Thanks for any help you can provide. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] s, CDR and NoCDR in v1.4.10.1
I am running 1.4.10.1. I have a macro that is called from default for a
certain extension (both below). I added NoCDR to s to try and stop
extra CDR records, but I am still getting them. Any idea how to stop them?
extensions.conf:
[macro-STDEXT]
exten =s,1,NoCDR()
exten =s,2,Dial(${ARG1},30,Tt)
exten =s,3,Goto(s-${DIALSTATUS},1)
exten =s-NOANSWER,1,Voicemail(${ARG2}|u)
exten =s-NOANSWER,n,Hangup
exten =s-BUSY,1,Voicemail(${ARG2}|u)
exten =s-BUSY,n,Hangup
exten =s-CONGESTION,1,Voicemail(${ARG2}|u)
exten =s-CONGESTION,n,Hangup
exten =s-CHANUNAVAIL,1,Voicemail(${ARG2}|u)
exten =s-CHANUNAVAIL,n,Hangup
[default]
exten =6080,1,Macro(STDEXT,SIP/6080,6080)
Here is an example. I am getting an 's' CDR with No Answer and then an
Answered CDR in default context:
6463,6463,s,default,SIP/6080-0861a5102007-12-04
11:49:30,,2007-12-04 11:49:39,9,0,NO
ANSWER,DOCUMENTATION,,1196790570.4260,
6463,6463,6080,default,SIP/206.190.240.9-082edc08,SIP/6080-086234e0,Dial,SIP/6080|30|Tt,
2007-12-04 11:49:30,2007-12-04 11:49:39,2007-12-04
11:49:44,14,5,ANSWERED,DOCUMENTATION,,1196790570.4259,
If I don't answer, I still get an 's' CDR with No Answer. Any ideas how
to stop that? Thanks.
Peder
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
Michael Melia Jr. wrote: Thanks for the suggestions so far. I don't like the idea that I have to give full control with OutCall but it seems to be the case with most of the solutions out there. I have downloaded and tested OutCall on Windows Vista and Outlook 2007. It doesn't seem to work 100% with Outlook 2007. Program looks promising but needs some revision for latest version of Outlook. It seems a lot of the issues I saw have been posted to OutCalls forum and they are working on the Outlook 2007 with Exchange integration. Maybe 1.5 will be what I am looking for. I am going to test on XP and Outlook 2003 later. I have not tried Activa TSP yet. Anyone have any feedback on that? I tried Activa TSP this afternoon with Outlook 2007 (on Windows XP). While the call I tried worked, the dialer came up with an error that said it must be restarted. If that's how it operates with Outlook 2007, I'd say it's not usable. I have not tried with Outlook 2003, but since I need a working solution for both Outlook 2003 and 2007. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple contacts.
On Wed, 5 Dec 2007, Lacy Moore wrote: the one you are logged into. Same as Asterisk. I can carry a phone with me, and plug it in and access my Asterisk server. I can login using softphones. Whatever phone I am on will ring. Unless the reregistration interval is fairly frequent, because the phones are behind crappy NAT gateways with poor statekeeping for UDP pinholes, etc. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDm804B
what module does the TDM804B use/need? Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)
I don't think it ever gets that cold here in Australia. PaulH On Wed, 2007-12-05 at 21:24 -0500, Jon Pounder wrote: Quoting Paul Hales [EMAIL PROTECTED]: another option is use some sort of linux based device n770, or even an nslu2, and program a sip client to behave however you like, then just fit the thing with a usb based speaker/mic. actually, a gamepad, speaker and mic with an n770 behind a piece of glass would make a pretty damn nice doorphone / look up the person in a directory kind of application. one thing to keep in mind though - lcd screens will actually freeze and crack if they get cold enough so make sure it says on all the time or heat the case if you do something like that and its actually outside the building itself. I have seen a beta-level unit that also supported POE. With regards to non-beta hardware, standard analog doorphones work pretty well with Linksys SPA units. PaulH On Tue, 2007-12-04 at 02:53 -0500, Nick Seraphin wrote: On a similar note... has anyone ever seen a SIP-based door intercom unit? Functionality I'm looking for is... basically an outdoor rated weather resistant speaker with 1 button and microphone, when the button is pressed, it dials a specified SIP extension. Likewise, from the Asterisk box, someone can call a SIP extension and the call goes to the intercom speaker so you can initiate a conversation with the person at the door if they just rang the bell but didn't push the intercom button. Preferably something with power over ethernet support. Thanks, -- Nick On Tue, 4 Dec 2007, Doug Meredith wrote: I have searched for this without much luck. I want to be able to send public-address-like notices over VoIP phones. The LinkSys SPA-941 auto-answer support comes close to working, except that if you are currently in a call it places that call on hold without warning. I'm willing to consider a more expensive phone to solve the problem if I have to. Thanks for any help you can provide. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Intercom? (was: Re: Phonewith public address functionality)
Lol, not even in Melbourne huh - BTW it's snowing here in NY again tonight :) Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 05, 2007 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Door Intercom? (was: Re: Phonewith public address functionality) I don't think it ever gets that cold here in Australia. PaulH On Wed, 2007-12-05 at 21:24 -0500, Jon Pounder wrote: Quoting Paul Hales [EMAIL PROTECTED]: another option is use some sort of linux based device n770, or even an nslu2, and program a sip client to behave however you like, then just fit the thing with a usb based speaker/mic. actually, a gamepad, speaker and mic with an n770 behind a piece of glass would make a pretty damn nice doorphone / look up the person in a directory kind of application. one thing to keep in mind though - lcd screens will actually freeze and crack if they get cold enough so make sure it says on all the time or heat the case if you do something like that and its actually outside the building itself. I have seen a beta-level unit that also supported POE. With regards to non-beta hardware, standard analog doorphones work pretty well with Linksys SPA units. PaulH On Tue, 2007-12-04 at 02:53 -0500, Nick Seraphin wrote: On a similar note... has anyone ever seen a SIP-based door intercom unit? Functionality I'm looking for is... basically an outdoor rated weather resistant speaker with 1 button and microphone, when the button is pressed, it dials a specified SIP extension. Likewise, from the Asterisk box, someone can call a SIP extension and the call goes to the intercom speaker so you can initiate a conversation with the person at the door if they just rang the bell but didn't push the intercom button. Preferably something with power over ethernet support. Thanks, -- Nick On Tue, 4 Dec 2007, Doug Meredith wrote: I have searched for this without much luck. I want to be able to send public-address-like notices over VoIP phones. The LinkSys SPA-941 auto-answer support comes close to working, except that if you are currently in a call it places that call on hold without warning. I'm willing to consider a more expensive phone to solve the problem if I have to. Thanks for any help you can provide. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)
Why not an ATA that has an FXS port with an autoanswer doorbox like this one: http://www.vikingelectronics.com/products/view_product.php?pid=428 On Dec 5, 2007 9:48 PM, Paul Hales [EMAIL PROTECTED] wrote: I don't think it ever gets that cold here in Australia. PaulH On Wed, 2007-12-05 at 21:24 -0500, Jon Pounder wrote: Quoting Paul Hales [EMAIL PROTECTED]: another option is use some sort of linux based device n770, or even an nslu2, and program a sip client to behave however you like, then just fit the thing with a usb based speaker/mic. actually, a gamepad, speaker and mic with an n770 behind a piece of glass would make a pretty damn nice doorphone / look up the person in a directory kind of application. one thing to keep in mind though - lcd screens will actually freeze and crack if they get cold enough so make sure it says on all the time or heat the case if you do something like that and its actually outside the building itself. I have seen a beta-level unit that also supported POE. With regards to non-beta hardware, standard analog doorphones work pretty well with Linksys SPA units. PaulH On Tue, 2007-12-04 at 02:53 -0500, Nick Seraphin wrote: On a similar note... has anyone ever seen a SIP-based door intercom unit? Functionality I'm looking for is... basically an outdoor rated weather resistant speaker with 1 button and microphone, when the button is pressed, it dials a specified SIP extension. Likewise, from the Asterisk box, someone can call a SIP extension and the call goes to the intercom speaker so you can initiate a conversation with the person at the door if they just rang the bell but didn't push the intercom button. Preferably something with power over ethernet support. Thanks, -- Nick On Tue, 4 Dec 2007, Doug Meredith wrote: I have searched for this without much luck. I want to be able to send public-address-like notices over VoIP phones. The LinkSys SPA-941 auto-answer support comes close to working, except that if you are currently in a call it places that call on hold without warning. I'm willing to consider a more expensive phone to solve the problem if I have to. Thanks for any help you can provide. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Running AGI script if condition met?
Hello
Some of our customers call with CID blocked. I'd like to save
those numbers into a SQLite database using a command-line PHP script,
so that I can...
1. Edit the CID name through a PHP web script which will just list all
the customers in the database who have a phone number but no CID name
set
2. Look up those customers' e-mail address from this database, and
send them an e-mail telling them that, if they're tired of having to
input their CID number every time they call, they should contact
whoever handles their PBX, so that it no longer blocks their number in
outgoing calls.
== Here's the command-line PHP script:
#!/usr/bin/php
?php
$fh = fopen('/root/output.txt', 'w');
fwrite($fh, Received . $argv[0]);
fclose($fh);
?
== Here's extensions.conf:
exten = 777,1,Set(CALLERIDNUM=1234567890)
exten = 777,n,ExecIf($[${LEN(${CALLERIDNUM})} =
10],AGI(/root/dummy.php),${CALLERIDNUM})
==... and here's the console :-/
-- Executing [EMAIL PROTECTED]:1] Set(SIP/9001-088aa918,
CALLERIDNUM=1234567890) in new stack
-- Executing [EMAIL PROTECTED]:2] ExecIf(SIP/9001-088aa918,
1|AGI(/root/dummy.php)|1234567890) in new stack
[Dec 6 04:40:46] WARNING[12293]: app_exec.c:186 execif_exec: Count
not find application! (AGI(/root/dummy.php))
=
It doesn't look like ExecIf() is the right way to have Asterisk run an
AGI script conditionnally. What would be the right way to do this?
Thank you.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Soundpoint (NO LINE)
At 19:19 12/5/2007, Ricardo Melendez wrote: Hi, I have just configure a Soundpoint 550 to work with Asterisk, it appear Registered to the asterisk server, and appear in asterisk console with SIP SHOW PEERS, and can receive calls, but when I try to dial, it launch a tone as if not line to dial, also can not stream audio to the other end when answer the received call. I have read the documentation but is very confuse. This phone have a Outbound Proxy (where I put the asterisk IP and port 5060) And 4 lines (where I configure the username/password and the Servers) Anyone can help me. Thanks in Advance. Atte. Ricardo Melendez What is extension number? What is the output of: sip show peer (extension number?) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running AGI script if condition met?
Vincent wrote:
exten = 777,n,ExecIf($[${LEN(${CALLERIDNUM})} =
10],AGI(/root/dummy.php),${CALLERIDNUM})
The line break is not a good idea.
It doesn't look like ExecIf() is the right way to have Asterisk run an
AGI script conditionnally. What would be the right way to do this?
Wrong syntax.
ExecIf(expr|app|data)
So:
ExecIf($[${LEN(${CALLERIDNUM})} = 10],AGI,/root/dummy.php)
Not sure about more than one argument. Maybe
ExecIf($[${LEN(${CALLERIDNUM})} = 10],AGI,/root/dummy.php,${CALLERIDNUM})
or
ExecIf($[${LEN(${CALLERIDNUM})} = 10],AGI,/root/dummy.php|${CALLERIDNUM})
Asterisk's syntax is strange sometimes ...
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running AGI script if condition met?
On Thu, 06 Dec 2007 05:11:24 +0100, Philipp Kempgen [EMAIL PROTECTED] wrote: The line break is not a good idea. It's not in the script, just my news reader :-) Not sure about more than one argument. Maybe Both work. Thanks a lot! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk server and DSCP QOS
Hi! Steve Johnson wrote: The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. If you are using VLAN, than you also look at new options in trunk cos_sip and cos_audio to set 802.1p. (If you run Linux). It will help with QoS too. -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can Asterix seperate the signalling and the media ip's with Quintum
New to Asterix and perhaps someone can help. The plnned configuration is that the Quintums are to register to the Asterix and the signalling to be handled by the Asterix but the media (G 729 code) to be directed to the service provider. Thanks Shaun ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 on wrong bus
Mark, This is the results [EMAIL PROTECTED] ~]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Pentium(R) 4 CPU 3.00GHz stepping: 1 cpu MHz : 2993.146 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 1 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe constant_tsc pni monitor ds_cpl cid xtpr bogomips: 5991.35 processor : 1 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Pentium(R) 4 CPU 3.00GHz stepping: 1 cpu MHz : 2993.146 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 1 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe constant_tsc pni monitor ds_cpl cid xtpr bogomips: 5985.45 On 11/28/07, broadband Voice [EMAIL PROTECTED] wrote: Hi, Can anyone assist me in resolving this problem? I installed the G729 on a 32 and just found out that the server is 64. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
