Re: [asterisk-users] Anybody have the opvx1200.c driver?
On Tue, Sep 26, 2006 at 08:43:11PM -0700, Nick Ellson wrote: > > The link is not working at OpenVox. There's a "download" link in the bottom of the page, that leads to: http://www.openvox.com.cn/members_downloads.php . That page has the "A1200P device driver" as a download item (not just for members). That page also reads: if you are using pop-up block tools, such as google toolbar, please close the pop-up block function before download. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Digium G.729 codec binaries updated for Asterisk 1.4 beta
On 2006-09-23 12:43:32 -0700, "Kevin P. Fleming" <[EMAIL PROTECTED]> said: - Matt Riddell (IT) <[EMAIL PROTECTED]> wrote: Also, are you referring to newer ones than the 1.4 downloads that were available a couple of days ago or do you mean people running the 1.2 versions? The versions that were initially posted as compatible with Asterisk 1.4 became incompatible just before beta2 was released, so these versions are compatible with beta2. How about some PowerPC love? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: e911
On 2006-09-24 17:51:51 -0700, "Tom Lynn" <[EMAIL PROTECTED]> said: I'm keeping my Qwest line for this purpose. Me too, but I hate paying them every month! I also do terminate some locals calls that way though... Also if all the power goes off this might still work ;~) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Two things come to mind. First, (I'm not familiar with the SonicWall, so this may be way off), could it have suddently decided that your Voip provider's IP address is a threat? From what I understand, Cisco uses some technology such as this. If it thinks there is a threat, it starts blocking things. Maybe try rebooting the firewall. I read that somewhere in the Cisco mountain of documents that it doesn't store this information permanently, just in RAM, and it gets reset upon reboot. Second, could your ISP have started blocking Voip packets? I imagine some providers who sell Voip services are going to start blocking and/or dropping packets. I can't be sure of this, but I never was able to have a Voip call last more than about 1 minute on Time Warner cable. My whole connection would drop. Switching the cat5 cable from Time Warner to my DSL provider (all other things being the same), I get calls that last as long as I want. Could be something to check into. If nothing changed on your end, it sounds like something changed between you and your Voip provider. Getting your ISP to admit to it, could be somewhat of a challenge. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change pager notification message
Look for pagerbody and pagersubject. On 9/26/06, Mike Diehl <[EMAIL PROTECTED]> wrote: Hi all.I currently get an alpha-page via email from Asterisk when I get a newvoicemail message. This is in ADDITION to getting the message emailed to me. I can iss in voicemail.conf that I can change the text of the email message,subject, and sender.But, how do I change the text of the alpha-page?TIA,Mike Diehl.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreI'm the guy that doesn't give a damn about anyone's problems but my own... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to change pager notification message
It's in voicemail.conf as well. On 9/26/06, Mike Diehl <[EMAIL PROTECTED]> wrote: Hi all.I currently get an alpha-page via email from Asterisk when I get a newvoicemail message. This is in ADDITION to getting the message emailed to me. I can iss in voicemail.conf that I can change the text of the email message,subject, and sender.But, how do I change the text of the alpha-page?TIA,Mike Diehl.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreI'm the guy that doesn't give a damn about anyone's problems but my own... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I doubt it...
I didn't see it as making fun of anyone. I, for one, was curious about it. I suspected it was some type of translation issue, whether it was a word in another language that doesn't translate or what. I know there are many concepts in English and in other languages that just doesn't translate correctly. I can't imagine how any software could translate all the different English dialects, so I'm sure translators have problems from other languages. On 9/26/06, Benjamin Jacob <[EMAIL PROTECTED]> wrote: Jay R. Ashworth wrote:>On Tue, Sep 26, 2006 at 05:33:15PM -0500, DiegoF wrote:>> >> hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan,>> />> hello to all, I have a doubt, ye I have solved some but others arrive, good >>*Oh*.>>*That's* where all these non-native English speakers are coming up with>"doubt". Someone's translator doesn't have an idiom for "I have an >inquiry".>>Eeediots.>>Easy.. Easy Jay!! easy duz it!You can't *expect* all to be native English speakers over here, oranywhere, for that matter. And am sure, he won't have a translator or a dictionary next to him, whenever he posts on this list.As long as ppl are harmless, are talking asterisk, are making sense,arn't cussing you out, its A-OK!!n besides, there isn't a Correct-English-talkers-only clause over here, I guess. Imagine a spanish-only (was that guy spanish??or mexican?baah.. all sound same) world and you would be sending "doubts" across aswell, with a Spanish Jay calling you an eeedioto!!live n let live. or over here, lets make it as "Asterisk and let Asterisk". now that i've flung my two cents, lets start a flame ;-)cheerz- Ben.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy MooreI'm the guy that doesn't give a damn about anyone's problems but my own... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] señalizacion te110p, signaling te110p
Maybe you could try an asterisk forum in spanish in order to get better results using your native language. DiegoF escribió: hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, bueno como habia dicho quiero conectar una pbx a una te110p, la pbx me ofrece señalizacion r2 europea en cable rj45 o coaxial. ese tipo de señalización me sirve para la tarjeta te110p, ademas, alguno de esos dos tipos de conexiones me sirven o tengo que comprar algun adaptador. vi algo que tenia que usar un balum, es necesario para cualquiera de las dos conexiones?. cual tipo de conexioon me recomiendan mas? necesito saber algo mas sobre la pbx para configurar en la te110p? atentamente diego fernando güiza arce / hello to all, I have a doubt, ye I have solved some but others arrive, good since te110p had said I want to connect a PBX to one, the PBX offers señalizaciòn to me r2 European in cable rj45 or coaxial that type of signaling is used for the card te110p to me, in addition, some of those two types of connections serves to me or I must buy some adapter. I saw something that tapeworm that to use a balum, is necessary for anyone of the two connections. as type of connection they recommend to me but? I need to know something but on the PBX to form in te110p? kindly diego fernando güiza arce // -- // DiegoF // // Dichosos aquellos que no esperan nada de la vida, porque nunca seran defraudados // // Se han fijado que cuando estan solos...no hay nadie??? // // Cada vez que me siento a pensar, lo unico que consigo es sentarme. // ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I doubt it...
Jay R. Ashworth wrote: On Tue, Sep 26, 2006 at 05:33:15PM -0500, DiegoF wrote: hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, / hello to all, I have a doubt, ye I have solved some but others arrive, good *Oh*. *That's* where all these non-native English speakers are coming up with "doubt". Someone's translator doesn't have an idiom for "I have an inquiry". Eeediots. Easy.. Easy Jay!! easy duz it! You can't *expect* all to be native English speakers over here, or anywhere, for that matter. And am sure, he won't have a translator or a dictionary next to him, whenever he posts on this list. As long as ppl are harmless, are talking asterisk, are making sense, arn't cussing you out, its A-OK!! n besides, there isn't a Correct-English-talkers-only clause over here, I guess. Imagine a spanish-only (was that guy spanish??or mexican? baah.. all sound same) world and you would be sending "doubts" across as well, with a Spanish Jay calling you an eeedioto!! live n let live. or over here, lets make it as "Asterisk and let Asterisk". now that i've flung my two cents, lets start a flame ;-) cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
(GOT IT) Re: [asterisk-users] Anybody have the opvx1200.c driver?
Thanks all, I have it now :) -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. On Tue, 26 Sep 2006, Nick Ellson wrote: The link is not working at OpenVox. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING: chan_sip.c add_realm_authentication: ???
When I reloaded my asterisk I saw these lines, which I have noticed before: [Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 797 [Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 822 [Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 847 [Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 872 [Sep 27 11:46:11] == Parsing '/etc/asterisk/sip_notify.conf': [Sep 27 11:46:11] Found What does it mean? Should I care? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anybody have the opvx1200.c driver?
The link is not working at OpenVox. Nick -- Nick Ellson CCDA, CCNP, CCSP, CCAI, MCSE 2000, Security+, Network+ Network Hobbyist, VFR Private Pilot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop wrote: Hi all I didn't change anything that's my point It has be running and working just fine then at 4:32 pm yesterday I could not make or recieve VoIP calls via our VoIP Provider They say the Invite packet was being rejected and thus there was no "real" connection even though SIP SHOW PEERS has us registered They also say it's due to the Sonicwall which has changed port assignments and thus blocking ports. I see in the Sonicwall log UDP Packet Dropped with the Providers IP Address but it talks about port 36612 which is not SIP They say along with the log that SIP is using 36612 why when all the VoIP SIP setting are enabled/configured and SIP is packet forwarded to the Asterisk Box due to Sonicwall NAT Are you sure that you're not confusing the SOURCE vs DESTINATION port? Your system would send a sip packet to your provider with a destination port of udp/5060, but your source port can be anything greater then 1024. (That's likely to be 36612 in your notes, above.) Your provider would reverse those in its response, sending their packet to the destination port of udp/ (the same greater then 1024 mentioned above), and a source port of udp/5060. That's just standard IP stuff. The nat function within the firewall keeps track of every udp and tcp conversation by building a table entry that includes source IP and source port (associated with the internal lan device that created the packet), and a destination IP and destination port (associated with your provider's device. That table entry is constantly referred to for every packet that passes through the firewall in either direction, translating private addresses into public addresses, etc. If the conversation is "udp" based, that table entry will timeout (and disappear) after some period of time. I don't recall what the default sonicwall timeout value happens to be, but its typically some number of low minutes (as opposed to low number of seconds). If the conversation is "tcp" based, that table entry will disappear when the tcp session is closed by the end devices. I can only guess that a tcp timeout value exists as well, however it would oriented around timing out a table entry where the end devices mysteriously disappeared (without closing the tcp session). Sonicwall sells their products with 10 user, 25 user, and other limits that would imply the above nat table size might have limits (or changes) when that maximum is reached. Are you sure you've not exceeded the license limit associated with your sonicswall? Sonicwall also has a problem handling udp packets that are greater in size then 1458 bytes (I think I have that value correct) when its wan interface is configured for PPPoE. Packets larger then that value are simply dropped on the floor, and no log entries are created to hint that has happened. Are you using PPPoE? Finally, sonicwall has implemented some sort of sip fixup that attempts to analyze the contents of a sip packet to determine which udp ports are to be used for rtp packets. I wouldn't think this function would have any impact in your case since it sounds like the problem is sip oriented and not rtp oriented. You could turn that option off just to ensure it isn't the problem. To diagnose this any further really requires a packet sniff (eg, ethereal) from the outside edge of the firewall, along with an asterisk 'sip debug'. That would help determine what might be happening in terms of port mapping, etc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue information
Hi, When I issue a show queues command, it shows something below. 2600 has 0 calls (max unlimited) in 'leastrecent' strategy (1s holdtime), W:0, C:491, A:29, SL:100.2% within 120s Does anyone know any reference to explain the meaning of the about information? 1s holdtime: what is that holdtime stand for? W, C, A: What are they mean? BTW, does it also contain in the database if I am using ARA? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] max number of devices in hint
I have one extension that rings in many places. It has just come to my attention that I can only monitor 4 devices within a hint. Ex: exten => 132,hint,SIP/DEVA&SIP/DEVB&SIP/DEVC&SIP/DEVD if I add SIP/DEVF, DEVF is not monitored. Is anyone else monitoring more than 4 devices, and if so, what version are you running? I'm running 1.2.12.1. I thought at first this may be a phone issue, but by running SHOW HINTS on the CLI, it shows InUse using the first four devices. The fifth device shows IDLE when it is, in fact, in use. I also switched places for DEVD and DEVE thinking maybe it was something with DEVE. When I switch places, DEVE shows as expected and DEVD does not show. Any ideas? -- Lacy Moore ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to change pager notification message
Hi all. I currently get an alpha-page via email from Asterisk when I get a new voicemail message. This is in ADDITION to getting the message emailed to me. I can iss in voicemail.conf that I can change the text of the email message, subject, and sender. But, how do I change the text of the alpha-page? TIA, Mike Diehl. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
Steve Totaro wrote: I set caller ID to a unique identifier before sending to a transfer partner or overflow call center. This makes it much easier to match CDRs and get stats on the outcome of calls once they leave our center. It is a very valuable and legitimate use. Am I committing a crime? nah. We use and trust ANI, not caller ID although I think I read you can manipulate ANI if you have an SS7 link. I have yet to play with SS7. Thanks, Steve Totaro Steve, This is exactly what I am talking about. That is a very useful application of caller id manipulation. Why should you lose that useful feature because a few misguided people sometimes use it for nefarious purposes? -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
Kristian Kielhofner wrote: Jay R. Ashworth wrote: But gratuituously making easy something that very few people have a legitimate need to do, which undermines something that -- even if you do only make the resaonable assumption that you know which phone, and not which person, is calling -- is useful and productive... is probably a Bad Idea. Full disclosure notwithstanding. Cheers, -- jra jra, Sprint made the mistake. That is ridiculous... Caller ID has not been secure for a long time. If you think that it should be made secure now, you are out of touch with reality because that is NOT going to happen. It has been made easy. It is ubiquitous. Get over it :)! The only options now are to not trust caller id, ask more questions (i.e. get better identity systems and processes in place), and, as I said, enforce laws that we already have. I think you missed my point that setting caller id in a nefarious way is almost always used as a tool in an action that is already defined as a crime. The things you are talking about doing are already illegal - whether or not you are spoofing caller id. Granted, caller id does make it easier, but if we didn't have the ability to set caller id the crooks would still be scamming, harassing, etc just like they are now. They would just be using other tools to do it or make it easier for them. -- Kristian Kielhofner I set caller ID to a unique identifier before sending to a transfer partner or overflow call center. This makes it much easier to match CDRs and get stats on the outcome of calls once they leave our center. It is a very valuable and legitimate use. Am I committing a crime? nah. We use and trust ANI, not caller ID although I think I read you can manipulate ANI if you have an SS7 link. I have yet to play with SS7. Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Priority "n"
>
> How do I use priority "n" correct?
>
First, which version of * are you using? Hopefully something recent.
If you've got 1.2.x then you can use n and labels. Check this out:
http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities
Tinker with it - you'll be surprised at how easy it is!
-MC
> Here is the current example:
>
> exten => 615,1,Dial(${PHONE_615},60,tr)
> exten => 615,2,Voicemail,[EMAIL PROTECTED]
> exten => 615,103,Voicemail,[EMAIL PROTECTED]
>
> and:
> exten => 617,n+101,GotoIf($["${DIALSTATUS}" :
> "(CHANUNAVAIL|CONGESTION)"]?110:999)
> exten => 617,110, .
>
> exten => 617,999,hangup
>
>
> That would greatly help me to throw out the NoOp statements I have
> inserted over the time if I tested some parts, ..
>
> bye
>
> Ronald
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
Jay R. Ashworth wrote: But gratuituously making easy something that very few people have a legitimate need to do, which undermines something that -- even if you do only make the resaonable assumption that you know which phone, and not which person, is calling -- is useful and productive... is probably a Bad Idea. Full disclosure notwithstanding. Cheers, -- jra jra, Sprint made the mistake. That is ridiculous... Caller ID has not been secure for a long time. If you think that it should be made secure now, you are out of touch with reality because that is NOT going to happen. It has been made easy. It is ubiquitous. Get over it :)! The only options now are to not trust caller id, ask more questions (i.e. get better identity systems and processes in place), and, as I said, enforce laws that we already have. I think you missed my point that setting caller id in a nefarious way is almost always used as a tool in an action that is already defined as a crime. The things you are talking about doing are already illegal - whether or not you are spoofing caller id. Granted, caller id does make it easier, but if we didn't have the ability to set caller id the crooks would still be scamming, harassing, etc just like they are now. They would just be using other tools to do it or make it easier for them. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I doubt it...
On Tue, Sep 26, 2006 at 05:33:15PM -0500, DiegoF wrote: >hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, >/ >hello to all, I have a doubt, ye I have solved some but others arrive, good *Oh*. *That's* where all these non-native English speakers are coming up with "doubt". Someone's translator doesn't have an idiom for "I have an inquiry". Eeediots. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
On Tue, Sep 26, 2006 at 07:17:57PM -0400, Kristian Kielhofner wrote: > Quite frankly, it is not my fault that the general public and several > institutions like banks, etc have poorly implemented systems on > THEIR end that ASSUME that CNID is gospel and use it for all kinds > of authentication purposes. Why do telcos use the ANI for billing? > Because it is gospel, No it's not. Sprint, in the southwest, converted presented CNID to ANI and sent it along. Check the telecom archives. But this isn't germane. > and as long as they are sending out bills, it > always will be. If you need to authenticate based on phone number > (which is ridiculous anyways), check against the ANI. If you are a > legit institution that needs access to the ANI, you should have no > problem getting that sent down your PRI from your telco. Indeed. But that's not on point, either. > Obviously caller ID is a joke, and has been for some time. That > ship sailed long before you and I started talking about it on > Asterisk-Users. The more that people fall for invalid and spoofed > caller id the better for all of us. Standard practice and public > opinion need to be changed on this. I understand your point, but I'm of two minds on this, as I am on the current ATM password fracas, and for the same reasons. > I hate getting credit cards and > having to activate them from my "home phone number". It tells me that > my credit card has no understanding of security for my account. Too > bad that to make purchases in the 21st century you need a credit card, > and all banks and card issuers are equally stupid. Indeed it is. > Why not connect me to a human that asks me all kinds of questions? I > know they can do that because other banks (and credit bureaus, etc) > have access to that info and have those processes in place. Oh yeah, they can ask you *useful* questions. Like your mother's maiden name. And your SSN. :-) > Maybe if US Weekly does a few more stories about celebs like Paris > Hilton getting jacked by spoofed caller id popular opinion might be > changed. Until then... Indeed. > What is boils down to is personal responsibility and enforcement of > rules/laws that are already in place. Sure, I *COULD* drive 150mph on > almost any road, but we as a society already have laws in place like > speed limits that will punish me when I do. I am not forbidden from > buying a Porsche (or penalized for having one) just because it can go > 150mph. However, if I do, I'll go to jail. Precisely. You're saying that "not spoofing caller ID" is not part of the American Social Contract, then? > Likewise, if a predator scams someone, stalks them, etc because they > have access to caller id spoofing, lock them up for theft or stalking > (illegal in most states). Don't take away their PRI or the ability to > set CID and punish the rest of us in the process. I'm no lawyer, but > in Wisconsin (and probably other states) it is perfectly legal and > acceptable to set caller id to anything you please, as long as it is > not used to stalk, harass, defraud, etc. If you get busted doing that, > not only do you faces charges on the original crime (stalking, theft, > etc) you get another count added for faking caller id to do it. The need to send CNID not your own for non-nefarious purposes (see the HP pretexting scam, and if you *don't* think that's going to dribble over into telemarketers sending fake CNID, TCPA notwithstanding, then you're nuts) is rare enough that I have no problem requiring the telcos to get a signed agreement from clients to turn off the filters. > As a matter of fact, a less known fact is that if you use an FRS > (Family Radio Service) walkie-talkie (or police scanner) in the > commission of a crime, you just broke another (federal) law and can be > prosecuted for that. There are examples of laws like this all over the > place... Sure. But gratuituously making easy something that very few people have a legitimate need to do, which undermines something that -- even if you do only make the resaonable assumption that you know which phone, and not which person, is calling -- is useful and productive... is probably a Bad Idea. Full disclosure notwithstanding. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
Jay R. Ashworth wrote: On Tue, Sep 26, 2006 at 05:28:12PM -0400, Kristian Kielhofner wrote: 2) Get a telco that lets you set any CID. I don't know if I just look trustworthy or something, but I have had no problems whatsoever getting several LECs and CLECs in multiple states to let me set any CID I want. Looking at the other posts, it seems that some people have problems with that. I never considered it to be a big deal, just a cool privilege that you gain with a PRI... It seems that isn't the case with some telcos. And, um, perhaps that's not a bad thing? Y'all read this: http://lauren.vortex.com/archive/000154.html and then give some more thought to whether you *should* play games with CNID... even assuming that you can. And don't give me "policy's not my problem; I'm only concerned with mechanism"... that's what they said at Birkenau, too. Hitler. Godwin. :-) Cheers, -- jra jra, Quite frankly, it is not my fault that the general public and several institutions like banks, etc have poorly implemented systems on THEIR end that ASSUME that CNID is gospel and use it for all kinds of authentication purposes. Why do telcos use the ANI for billing? Because it is gospel, and as long as they are sending out bills, it always will be. If you need to authenticate based on phone number (which is ridiculous anyways), check against the ANI. If you are a legit institution that needs access to the ANI, you should have no problem getting that sent down your PRI from your telco. Obviously caller ID is a joke, and has been for some time. That ship sailed long before you and I started talking about it on Asterisk-Users. The more that people fall for invalid and spoofed caller id the better for all of us. Standard practice and public opinion need to be changed on this. I hate getting credit cards and having to activate them from my "home phone number". It tells me that my credit card has no understanding of security for my account. Too bad that to make purchases in the 21st century you need a credit card, and all banks and card issuers are equally stupid. Why not connect me to a human that asks me all kinds of questions? I know they can do that because other banks (and credit bureaus, etc) have access to that info and have those processes in place. Maybe if US Weekly does a few more stories about celebs like Paris Hilton getting jacked by spoofed caller id popular opinion might be changed. Until then... What is boils down to is personal responsibility and enforcement of rules/laws that are already in place. Sure, I *COULD* drive 150mph on almost any road, but we as a society already have laws in place like speed limits that will punish me when I do. I am not forbidden from buying a Porsche (or penalized for having one) just because it can go 150mph. However, if I do, I'll go to jail. Likewise, if a predator scams someone, stalks them, etc because they have access to caller id spoofing, lock them up for theft or stalking (illegal in most states). Don't take away their PRI or the ability to set CID and punish the rest of us in the process. I'm no lawyer, but in Wisconsin (and probably other states) it is perfectly legal and acceptable to set caller id to anything you please, as long as it is not used to stalk, harass, defraud, etc. If you get busted doing that, not only do you faces charges on the original crime (stalking, theft, etc) you get another count added for faking caller id to do it. As a matter of fact, a less known fact is that if you use an FRS (Family Radio Service) walkie-talkie (or police scanner) in the commission of a crime, you just broke another (federal) law and can be prosecuted for that. There are examples of laws like this all over the place... -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Priority "n"
How do I use priority "n" correct?
Here is the current example:
exten => 615,1,Dial(${PHONE_615},60,tr)
exten => 615,2,Voicemail,[EMAIL PROTECTED]
exten => 615,103,Voicemail,[EMAIL PROTECTED]
and:
exten => 617,109,GotoIf($["${DIALSTATUS}" :
"(CHANUNAVAIL|CONGESTION)"]?110:999)
exten => 617,110, .
exten => 617,999,hangup
That would greatly help me to throw out the NoOp statements I have
inserted over the time if I tested some parts, ..
bye
Ronald
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - alcatel
Nicolas, We use a TE110P from digium. We make the same procedures oriented in that website. the only change was in signaling as i've said previously. My alcatel aready have an E1 ISDN installed from local carrier. After asterisk is setup, we change cables from carrier to asterisk, and our span stay in green state. Wich pins of cable you use in ISDN cable ?? What is the result of zttools -v ??? After span configuration we have problemas making calls, se my post in other forum: http://forums.digium.com/viewtopic.php?t=9868&highlight=alcatel+4200 -- - Frederico Madeira [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] ICQ: 37152149 SKYPE: fred_madeira Registered GNU/Linux nº 206120 -- Powered by LINUX -- CCNA Em Ter, 2006-09-26 às 23:25 +0200, Nicolas Bocquet escreveu: Hello, We have test this configuration but we think it's a problem with the Alcatel. how are you doing to make the trunk between alcatel and Asterisk? We use a card PRA recommended by an Alcatel's technician and you? Thanks Nicolas On 9/26/06, Frederico Madeira <[EMAIL PROTECTED]> wrote: I'm trying the same in Alcatel 4200 and i solved changing ignaling from pri_cpe for pri_net. -- Frederico Madeira [EMAIL PROTECTED] 2006/9/26, Sylvain ZUCCA <[EMAIL PROTECTED]>: Hi, can you send logs from alcatel 4400 ? just log in with account mtcl and launch "t3" to see traces from the PBX Best Regards. 2006/9/26, et pourquoi pas ? epp <[EMAIL PROTECTED]>: Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). But I think there is a link with the fact that the digium card (110) is always yellow Do you have a idea for me ? Best regards, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: Esta é uma parte de mensagem assinada digitalmente ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream GXV 3000
Hi, Someone has a Grandstream GXV 3000 to run a small test with me? I have one GXV 3000 setup and I can’t get video from that videophone with Eyebeam. Best regards, Chris HARIGA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Context default & incoming ENUM
I want to make the context [default] as an alarm, for not having set-up correct. I am looking for a way to get incoming calls via ENUM or via names (e.g. sip:[EMAIL PROTECTED]) into a defined context. How can I do that? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] señalizacion te110p, signaling te110p
What a confused message, isn't it? As far as I could understand, if you're getting a RJ45 for conection, you won't need any kind of adaptor. For coaxial cable, you'll need a balun. That's all layer 1 talk - physic layer Yes, you need to know a lot more about your pbx to proceed with the connection to your * box(TE110P). []'s MM -Original Message- From: DiegoF <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Cc: Sent: Tue, 26 Sep 2006 17:33:15 -0500 Delivered: Tue, 26 Sep 2006 19:28:55 Subject:[asterisk-users] señalizacion te110p, signaling te110p hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, bueno como habia dicho quiero conectar una pbx a una te110p, la pbx me ofrece señalizacion r2 europea en cable rj45 o coaxial. ese tipo de señalización me sirve para la tarjeta te110p, ademas, alguno de esos dos tipos de conexiones me sirven o tengo que comprar algun adaptador. vi algo que tenia que usar un balum, es necesario para cualquiera de las dos conexiones?. cual tipo de conexioon me recomiendan mas? necesito saber algo mas sobre la pbx para configurar en la te110p? atentamente diego fernando güiza arce / hello to all, I have a doubt, ye I have solved some but others arrive, good since te110p had said I want to connect a PBX to one, the PBX offers señalizaciòn to me r2 European in cable rj45 or coaxial that type of signaling is used for the card te110p to me, in addition, some of those two types of connections serves to me or I must buy some adapter. I saw something that tapeworm that to use a balum, is necessary for anyone of the two connections. as type of connection they recommend to me but? I need to know something but on the PBX to form in te110p? kindly diego fernando güiza arce // -- // DiegoF // // Dichosos aquellos que no esperan nada de la vida, porque nunca seran defraudados // // Se han fijado que cuando estan solos...no hay nadie??? // // Cada vez que me siento a pensar, lo unico que consigo es sentarme. // E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l=1,1159311041.809165.19547.alcala.hst.terra.com.br,7527,Des15,Des15 --Original Message Ends-- -- Melcon Moraes <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Included context
Great idea but I am looking for more informative answer than 'might
accidentally happen something'.
But anyway here is the result for your pleasure:
calling with context1:
-- Executing NoOp("Zap/32-1", "CONTEXT IS: context1 DIAL") in new stack
-- Executing Wait("Zap/32-1", "1") in new stack
-- Executing Playback("Zap/32-1", "context1") in new stack
-- Executing Wait("Zap/32-1", "1") in new stack
-- Executing Hangup("Zap/32-1", "") in new stack
== Spawn extension (context1, s, 5) exited non-zero on 'Zap/32-1'
-- Executing NoOp("Zap/32-1", "CONTEXT IS: context1 HANGUP") in new stack
-- Hungup 'Zap/32-1'
calling with context2:
-- Executing NoOp("Zap/32-1", "CONTEXT IS: context2 DIAL") in new stack
-- Executing Wait("Zap/32-1", "1") in new stack
-- Executing Playback("Zap/32-1", "context2") in new stack
-- Executing Wait("Zap/32-1", "1") in new stack
-- Executing Hangup("Zap/32-1", "") in new stack
== Spawn extension (context2, s, 5) exited non-zero on 'Zap/32-1'
-- Executing NoOp("Zap/32-1", "CONTEXT IS: context2 HANGUP") in new stack
-- Hungup 'Zap/32-1'
calling with context4 (which includes context2):
-- Executing NoOp("Zap/32-1", "CONTEXT IS: context4 DIAL") in new stack
-- Executing Wait("Zap/32-1", "1") in new stack
-- Executing Playback("Zap/32-1", "context4") in new stack
-- Executing Wait("Zap/32-1", "1") in new stack
-- Executing Hangup("Zap/32-1", "") in new stack
== Spawn extension (context4, s, 5) exited non-zero on 'Zap/32-1'
-- Executing NoOp("Zap/32-1", "CONTEXT IS: context4 HANGUP") in new stack
So looks like ${CONTEXT} is equal with originating context.
bye,
Zsolt
On 9/26/06, C F <[EMAIL PROTECTED]> wrote:
How about trying Noop(CONTEXT IS: ${CONTEXT}) before you post it? but
now that you have posted just try it and report back.
On 9/26/06, Artifex Maximus <[EMAIL PROTECTED]> wrote:
> Hello,
>
> For example I have this dialplan:
>
> [context1]
> exten => s,1,Noop
> exten => s,n,Dial(...)
> exten => s,n,Playback(${CONTEXT})
> exten => s,n,Hangup
>
> [context2]
> include => context1
>
> [context3]
> include => context1
>
> Then I make dial-out call files with context2, context3, etc. What is
> the value of ${CONTEXT} in that case? Still context1 because it's
> physically there or context2, context3 because I am included from
> there (so in a way logically is there). I didn't find any exact answer
> that's why I'm asking here.
>
> bye,
> Zsolt
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with "Background" DTMF detection with A200D
Hi all,
I'm having trouble with Background DTMF detection, and would appreciate
any suggestions.
A call comes in to a Sangoma A200D PSTN line. A standard menu welcome
is used. Most of the time, callers have to wait until the message
completes in order to have their selection recognized. People end up
having to press the option number several times. Occasionally, you can
press the desired option digit during the message and it will be
selected right away while the Background message is still playing (this
is what I want all the time). Any suggestions?
Environment: Asterisk 1.2.10, zaptel-1.2.7, wanpipe-beta7-2.3.4.tgz
Machine has lots of horsepower: Pentium D 3.2 GHz, 2 GB RAM,
[general]
priorityjumping=no
autofallthrough=no
(...)
[from-pstn]
; Inbound calls from PSTN line
exten => s,1,NoOp(TIMESTAMP: ${TIMESTAMP})
exten => s,2,NoOp(CONTEXT: ${CONTEXT})
exten => s,3,NoOp(CALLERIDNUM: ${CALLERIDNUM})
exten => s,4,NoOp(CALLERIDNAME: ${CALLERIDNAME})
exten => s,n,Goto(mainmenu,s,1)
[mainmenu]
exten => s,1,NoOp(Main Menu)
exten => s,n,Wait,1
exten => s,n,Answer
exten => s,n,Set(TIMEOUT(digit)=5)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Playback(silence-1sec)
exten => s,n,Playback(silence-1sec)
exten => s,n,Background(mainmenu)
; Thank you for calling xxx.
;Please press 1 for AA;
;2 for BB;
;3 for CC;
;or 4 for DD.
;Press 0, or stay on the line for reception.
exten => 1,1,NoOp(Menu 1 - Dialing SIP/101 AA)
exten => 1,n,Dial(SIP/101,20,t)
exten => 1,n,Playback(silence-1sec)
exten => 1,n,Voicemail(u101)
exten => 1,n,Hangup
exten => 2,1,NoOp(Menu 2 - Dialing SIP/102 BB)
exten => 2,n,Dial(SIP/102,20,t)
exten => 2,n,Playback(silence-1sec)
exten => 2,n,Voicemail(u102)
exten => 2,n,Hangup
exten => 3,1,NoOp(Menu 1 - Dialing SIP/103 CC)
exten => 3,n,Dial(SIP/103,20,t)
exten => 3,n,Playback(silence-1sec)
exten => 3,n,Voicemail(u103)
exten => 3,n,Hangup
exten => 4,1,NoOp(Menu 1 - Dialing SIP/104 DD)
exten => 4,n,Dial(SIP/104,20,t)
exten => 4,n,Playback(silence-1sec)
exten => 4,n,Voicemail(u104)
exten => 4,n,Hangup
exten => 0,1,NoOp(Menu 0 - Dialing SIP/100)
exten => 0,n,Dial(SIP/100,20,t)
exten => 0,n,Playback(silence-1sec)
exten => 0,n,Voicemail(u100)
exten => 0,n,Hangup
exten => #,1,NoOp(Menu # - Access VOICEMAIL)
exten => #,n,Playback(silence-1sec)
exten => #,n,VoiceMailMain()
exten => #,n,Hangup
;
exten => t,1,NoOp(Menu t - Goto mainmenu,0,1)
exten => t,n,Goto(mainmenu,0,1)
;
exten => i,1,NoOp(Menu i - Playback pbx-invalid)
exten => i,n,Playback(pbx-invalid)
exten => i,n,Goto(mainmenu,s,1)
;end of [mainmenu]
;
In the zapata.conf file, the relevant parts are:
[trunkgroups]
[channels]
language=en
context=default
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
cidsignalling=bell
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
musiconhold=default
echocancel=yes
echocancelwhenbridged=yes
rxgain=3.0
txgain=0.0
immediate=no
faxdetect=no
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 1
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 2
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 3
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 4
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 5
group=1
signalling=fxs_ks
context=from-pstn
language=en
channel => 6
;---
Thanks for any ideas,
Alvin
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] señalizacion te110p, signali ng te110p
hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, bueno como habia dicho quiero conectar una pbx a una te110p, la pbx me ofrece señalizacion r2 europea en cable rj45 o coaxial. ese tipo de señalización me sirve para la tarjeta te110p, ademas, alguno de esos dos tipos de conexiones me sirven o tengo que comprar algun adaptador. vi algo que tenia que usar un balum, es necesario para cualquiera de las dos conexiones?. cual tipo de conexioon me recomiendan mas? necesito saber algo mas sobre la pbx para configurar en la te110p? atentamente diego fernando güiza arce/hello to all, I have a doubt, ye I have solved some but others arrive, good since te110p had said I want to connect a PBX to one, the PBX offers señalizaciòn to me r2 European in cable rj45 or coaxial that type of signaling is used for the card te110p to me, in addition, some of those two types of connections serves to me or I must buy some adapter. I saw something that tapeworm that to use a balum, is necessary for anyone of the two connections. as type of connection they recommend to me but? I need to know something but on the PBX to form in te110p? kindlydiego fernando güiza arce//-- // DiegoF Dichosos aquellos que no esperan nada de la vida, porque nunca seran defraudados Se han fijado que cuando estan solos...no hay nadie??? // // Cada vez que me siento a pensar, lo unico que consigo es sentarme. // ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] speaker phone echo
Try a different, (larger or smaller) room with different acoustical characteristics. You may be talking, the audio is transmitted from a primary source - you - but then it may pick up the reflections of your voice bouncing off of the walls in the room, and the phone may be picking that up as well. You may not be able to hear the difference, but your phone can. Also reduce TX gain on the phone. hth -Original Message-From: Christopher Corn [mailto:[EMAIL PROTECTED]Sent: Tuesday, September 26, 2006 3:47 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] speaker phone echo I'm having speaker phone echo issues with my grandstream phones 100. i understand that the echo'ing issue is only obvious because of the round trip latency and that traditional phone lines have echo's too but because there is such a slight delay, it can be mistaken for side tone, which is perfectly normal. I don't have any echo issues now on my phones, unless i turn on the speaker phone, then i get acoustic echos. anyone else having this problem? what are some things I can do to alleviate this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
On Tue, Sep 26, 2006 at 05:28:12PM -0400, Kristian Kielhofner wrote: > 2) Get a telco that lets you set any CID. I don't know if I just look > trustworthy or something, but I have had no problems whatsoever getting > several LECs and CLECs in multiple states to let me set any CID I want. > Looking at the other posts, it seems that some people have problems > with that. I never considered it to be a big deal, just a cool > privilege that you gain with a PRI... It seems that isn't the case with > some telcos. And, um, perhaps that's not a bad thing? Y'all read this: http://lauren.vortex.com/archive/000154.html and then give some more thought to whether you *should* play games with CNID... even assuming that you can. And don't give me "policy's not my problem; I'm only concerned with mechanism"... that's what they said at Birkenau, too. Hitler. Godwin. :-) Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Hi Francesco Yes it is SonicWALL Enhanced has an option called 'Persistent NAT'... Is it turned on? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: IAX2 & SIP Monitoring Solution for Asterisk
| Just wondering if anyone has come up with a reliable method for | remotely monitoring Asterisk boxes. | | I need to be able to check if Asterisk is actually providing service | (registering clients, processing calls), not just answering to pings. | | In the past I have used sipsak in a cron script to do SIP | registrations, but I haven't been able to find anything similar for | IAX2. Any thoughts? argus can monitor using both SIP and IAX. see: http://argus.tcp4me.com http://www.voip-info.org/wiki/view/Asterisk+monitoring ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mISDN, 2 Billion HFC ISDN cards, cannot dial or receive
I have two similar Billion HFC cards working on a bri-stuffed Asterisk 0.99 server which has run my phone system in the UK for 2 years without fault. The hardware is getting old, and I thought it wilse to upgrade at my own speed, rather than in response to a failure. I've installed Asterisk 1.2.12.1 via the Beronet "install-asterisk" script and it appears to have installed mISDN correctly too. Placing a call to ISDN I get silence for 6 seconds, and then the call is dropped. Placing an incoming call over the PSTN from my mobile phone, i get no acknowledgement from the Asterisk, and a continuous tone from the mobile. The mISDN log (log level 4) for the outgoing call is below. Does anyone have a similar system working? Does anyone know why nothing appears to happen? I'll post a solution when I get one... Thanks, Nigel. Tue Sep 26 21:48:04 2006: P[ 0] --> Group Call group: bri Tue Sep 26 21:48:04 2006: P[ 1] Group [bri] Port [1] Tue Sep 26 21:48:04 2006: P[ 1] portup:1 Tue Sep 26 21:48:04 2006: P[ 0] --> * NEW CHANNEL dad:02012345678 oad:(null) Tue Sep 26 21:48:04 2006: P[ 1] * Queuing chan 0x958c0a0 Tue Sep 26 21:48:04 2006: P[ 1] read_config: Getting Config Tue Sep 26 21:48:04 2006: P[ 1] config_jb: Called Tue Sep 26 21:48:04 2006: P[ 1] --> * CallGrp: PickupGrp: Tue Sep 26 21:48:04 2006: P[ 1] --> TON: Unknown Tue Sep 26 21:48:04 2006: P[ 1] --> LTON: Unknown Tue Sep 26 21:48:04 2006: P[ 1] --> CTON: Unknown Tue Sep 26 21:48:04 2006: P[ 1] * CALL: g:bri/02012345678 Tue Sep 26 21:48:04 2006: P[ 1] --> * dad:02012345678 tech:mISDN/1-u3 ctx:incoming Tue Sep 26 21:48:04 2006: P[ 1] --> * adding2newbc ext 02077347184 Tue Sep 26 21:48:04 2006: P[ 1] --> * adding2newbc callerid 31 Tue Sep 26 21:48:04 2006: P[ 1] update_config: Getting Config Tue Sep 26 21:48:04 2006: P[ 1] --> pres: 1 screen: 1 Tue Sep 26 21:48:04 2006: P[ 1] NO OPTS GIVEN Tue Sep 26 21:48:04 2006: P[ 1] I SEND:SETUP oad:31 dad:02012345678 pid:5 Tue Sep 26 21:48:04 2006: P[ 1] --> bc_state:BCHAN_CLEANED Tue Sep 26 21:48:04 2006: P[ 1] --> channel:0 mode:TE cause:16 ocause:16 rad: cad: Tue Sep 26 21:48:04 2006: P[ 1] --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0 Tue Sep 26 21:48:04 2006: P[ 1] --> caps:Speech pi:0 keypad: sending_complete:0 Tue Sep 26 21:48:04 2006: P[ 1] --> screen:1 --> pres:1 Tue Sep 26 21:48:04 2006: P[ 1] --> addr:0 l3id:50003 b_stid:0 layer_id:0 Tue Sep 26 21:48:04 2006: P[ 1] --> facility:FAC_NONE out_facility:FAC_NONE Tue Sep 26 21:48:04 2006: P[ 1] --> urate:0 rate:16 mode:0 user1:0 Tue Sep 26 21:48:04 2006: P[ 1] --> bc:953e504 h:0 sh:0 Tue Sep 26 21:48:04 2006: P[ 1] --> new_l3id 50004 Tue Sep 26 21:48:04 2006: P[ 1] Sending msg, prim:30580 addr:41000104 dinfo:50004 Tue Sep 26 21:48:04 2006: P[ 1] --> * SEND: State Dialing pid:5 Tue Sep 26 21:48:10 2006: P[ 1] handle_frm: frm->addr:42000103 frm->prim:3f182 Tue Sep 26 21:48:10 2006: P[ 1] --> lib: RELEASE_CR Ind with l3id:50004 Tue Sep 26 21:48:10 2006: P[ 1] --> lib: CLEANING UP l3id: 50004 Tue Sep 26 21:48:10 2006: P[ 1] empty_chan_in_stack: 255 Tue Sep 26 21:48:10 2006: P[ 1] $$$ CLEANUP CALLED pid:5 Tue Sep 26 21:48:10 2006: P[ 1] $$$ Already cleaned up bc with stid :0 Tue Sep 26 21:48:10 2006: P[ 1] I IND :CLEAN_UP oad: dad: pid:5 state:CALLING Tue Sep 26 21:48:10 2006: P[ 1] hangup_chan Tue Sep 26 21:48:10 2006: P[ 1] -> queue_hangup Tue Sep 26 21:48:10 2006: P[ 1] release_chan: bc with l3id: 50004 Tue Sep 26 21:48:10 2006: P[ 1] * RELEASING CHANNEL pid:5 ctx:incoming dad:02077347184 oad:02077347184 state: CALLING Tue Sep 26 21:48:10 2006: P[ 1] --> * State Down Tue Sep 26 21:48:10 2006: P[ 1] --> Setting AST State to down ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop wrote: Hi all I didn't change anything that's my point It has be running and working just fine then at 4:32 pm yesterday I could not make or recieve VoIP calls via our VoIP Provider They say the Invite packet was being rejected and thus there was no "real" connection even though SIP SHOW PEERS has us registered They also say it's due to the Sonicwall which has changed port assignments and thus blocking ports. I see in the Sonicwall log UDP Packet Dropped with the Providers IP Address but it talks about port 36612 which is not SIP They say along with the log that SIP is using 36612 why when all the VoIP SIP setting are enabled/configured and SIP is packet forwarded to the Asterisk Box due to Sonicwall NAT Now I'm trying to find out why and how to correct this. Thanks all Barry Barry, First of all, devices like SonicWall drive me (and a lot of other people) crazy because of all of their "protocol helpers" that seems to break things more often than fix them. FTP with sonicwall was always a classic example - their active FTP helper was totally useless for a while. It seems that if you are totally clueless they offer some degree of help. If you know what you are doing, they get in the way. Anyways, as it is now try to enable sip debugging on the Asterisk console: "sip debug" This will show you all of the SIP messages to/from the Asterisk system. Try to make a call and see if the INVITE makes it to Asterisk (your console will print it out). If it makes it, look for "found peer xxx" shortly after the INVITE. It should match the name of your supposed incoming peer. Then it should match the correct context and do it's thing. However, your situation probably won't be that simple... My guess is the INVITE gets there, but the From/To/URI/something else are being mangled in the SIP request, so Asterisk doesn't know which peer in sip.conf to match them to. There was a doc on the WIKI (I can't find it right now) that describes how Asterisk matches incoming SIP requests to peers, contexts, etc. I'm pretty sure it works like this: 1) Try to match From: username to user= line in sip.conf [peer/user/friend] section. 2) Try to match source IP address of SIP request to known peer IP address. I.e, if you have host=sip.krisk.org in a [peer] section and the invite comes from 169.207.1.3 (which is sip.krisk.org), it uses that peer entry and corresponding context, etc. 3) If it doesn't match either, it goes into the context specified in [general] in sip.conf. As I said before, my guess is the SonicWall is getting fancy on you and breaking these otherwise reasonable sane methods. "sip debug" is your friend here. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] speaker phone echo
I'm having speaker phone echo issues with my grandstream phones 100. i understand that the echo'ing issue is only obvious because of the round trip latency and that traditional phone lines have echo's too but because there is such a slight delay, it can be mistaken for side tone, which is perfectly normal. I don't have any echo issues now on my phones, unless i turn on the speaker phone, then i get acoustic echos. anyone else having this problem? what are some things I can do to alleviate this?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
Shawn Kelley wrote: Hi all, I've searched around and haven't found much of an answer to my issue. Any advice from you would be appreciated. Problem: Need to take an inbound call from our PRI and forward it to another PSTN user via the PRI, sending the original callers id with it. I know this can be done since we currently use an 800 service that does it. You call the 800 number; they answer and put you on hold. They then outcall to the pstn numbers we have defined and the incoming call shows up with the original callers CID, we answer and have options to accept or reject the call. So I know the 800 provider is staying in the middle of the call and not just performing a redirect to us. I've tried the various CID settings in Asterisk, but am not able to use anything but our DID numbers for our outbound caller id. My telco has been unresponsive to this issue. Does anyone know if it's possible with a PRI or do you have to have some other type of PSTN connection such as SS7? Thanks!! --Shawn Shawn, 1) When a call comes in, put the original CALLERID(number) into a variable. This way, if you mess with the real CALLERIDNUM through your dialplan you can always set it back. I like to use KKFROMCID to make sure that no scripts, Asterisk, etc mess with my original CID! 2) Get a telco that lets you set any CID. I don't know if I just look trustworthy or something, but I have had no problems whatsoever getting several LECs and CLECs in multiple states to let me set any CID I want. Looking at the other posts, it seems that some people have problems with that. I never considered it to be a big deal, just a cool privilege that you gain with a PRI... It seems that isn't the case with some telcos. 3) I don't know if this works or not, but I could swear that there is a redirect possible on PRI (similar to SIP 302). I don't know if app_transfer (and your telco) support it, but it would be really cool because it would save you in terms of the number of used channels. (Using 0 channels instead of 2). Check this thread: http://lists.digium.com/pipermail/asterisk-users/2003-May/004594.html -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 & SIP Monitoring Solution for Asterisk
Greeting Everyone, Just wondering if anyone has come up with a reliable method for remotely monitoring Asterisk boxes. I need to be able to check if Asterisk is actually providing service (registering clients, processing calls), not just answering to pings. In the past I have used sipsak in a cron script to do SIP registrations, but I haven't been able to find anything similar for IAX2. Any thoughts? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - alcatel
Hello, We have test this configuration but we think it's a problem with the Alcatel.how are you doing to make the trunk between alcatel and Asterisk?We use a card PRA recommended by an Alcatel's technician and you? ThanksNicolasOn 9/26/06, Frederico Madeira <[EMAIL PROTECTED]> wrote: I'm trying the same in Alcatel 4200 and i solved changing ignaling from pri_cpe for pri_net.-- Frederico Madeira [EMAIL PROTECTED] 2006/9/26, Sylvain ZUCCA <[EMAIL PROTECTED]>: Hi, can you send logs from alcatel 4400 ? just log in with account mtcl and launch "t3" to see traces from the PBX Best Regards. 2006/9/26, et pourquoi pas ? epp <[EMAIL PROTECTED]>: Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). But I think there is a link with the fact that the digium card (110) is always yellow Do you have a idea for me ? Best regards, Thomas ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
On Tue, September 26, 2006 22:21, Barry Fawthrop wrote: > Hi all > > I didn't change anything that's my point > It has be running and working just fine then at 4:32 pm yesterday I > could not make or recieve VoIP calls via our VoIP Provider > They say the Invite packet was being rejected and thus there was no > "real" connection even though SIP SHOW PEERS has us registered > > They also say it's due to the Sonicwall which has changed port > assignments and thus blocking ports. > I see in the Sonicwall log UDP Packet Dropped with the Providers IP > Address but it talks about port 36612 which is not SIP > > They say along with the log that SIP is using 36612 why when all the > VoIP SIP setting are enabled/configured and SIP is packet forwarded to the > Asterisk Box due to Sonicwall NAT > > > Now I'm trying to find out why and how to correct this. > > > Thanks all > Barry > > SonicWALL Enhanced has an option called 'Persistent NAT'... Is it turned on? -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE406P not working on Intel D101Ggc motherboard.
On Tue, Sep 26, 2006 at 03:03:23PM -0400, ?lvaro Palma wrote: > I recently moved a TE406P card from an Intel D865GBF motherboard (where > it worked fine), to an Intel D101Ggc card, and now I can't get the spans > to got up correctly. All I get is an endless burst of: As much of a pain as it is, I always as, in such circumstances: did you put it back on the original working mobo? Does it still work? They don't call it provocative maintenance for nothing. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Hi all I didn't change anything that's my point It has be running and working just fine then at 4:32 pm yesterday I could not make or recieve VoIP calls via our VoIP Provider They say the Invite packet was being rejected and thus there was no "real" connection even though SIP SHOW PEERS has us registered They also say it's due to the Sonicwall which has changed port assignments and thus blocking ports. I see in the Sonicwall log UDP Packet Dropped with the Providers IP Address but it talks about port 36612 which is not SIP They say along with the log that SIP is using 36612 why when all the VoIP SIP setting are enabled/configured and SIP is packet forwarded to the Asterisk Box due to Sonicwall NAT Now I'm trying to find out why and how to correct this. Thanks all Barry Rich Adamson wrote: Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Yes, have multiple clients with asterisk behind a sonicwall. I don't understand from your wording if you mean a voip connection suddenly changed from dup/5060, or, did you change the asterisk system to use some other udp port. The sonicwall does have an option to support sip (udp/5060), but I've not had to use it on anything that we've worked with. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
On Tue, Sep 26, 2006 at 12:49:34PM -0500, Lacy Moore - Aspendora wrote: >Yes, it is possible. But, your Telco has to support this. Your Telco has >to give you the ability to set your caller ID. Some providers (and it >sounds like yours may be one of them) only allow you to use numbers which >you are authorized to use (such as your DIDs). Specifically, carriers who permit you to connect using a technology which allows you to send originating CNID (which is basically limited to ISDN at the moment, I believe) *are supposed to* filter the CNID you present before passing it along (I believe this to be in Part 68, but can't cite it), but not all of them do. In the past, 5ESS's automatically filtered, and DMS-100's automatically didn't, though either could -- I think -- be datafilled on a trunkgroup basis to work the other way. In the OP's situation, if his carrier doesn't already forward the CNID he supplies them, then he'll likely have to sign something with the to get authorization to do it. Or, like someone said, pretext it. Oh, my; that's a bad word this year. :-) And it's not real rugged either. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk to cell phone network
On 18:32, Tue 26 Sep 06, Andrea Spadaccini wrote: > Well, how does Asterisk interact with those devices? Is there a > chan_gsm_pci? It's using chan_zap. junghanns.net created an extra zap driver for it, same as with their quad/octobri and zap_hfc stuff. So asterisk will see it as Zap/ -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer afficionados are both called users?" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4 mohsuggest
Ok, so does anyone know who the contributor of the new moh code is into Asterisk 1.4? I'll email them directly. Doug. > -Original Message- > From: Douglas Garstang > Sent: Tuesday, September 26, 2006 8:31 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Asterisk 1.4 mohsuggest > > > I'm trying to get moh working correctly in Asterisk 1.4. A > complete lack of documentation isn't helping much. > > I have this in sip.conf: > > [3254101] > type=friend > ... > mohsuggest=class1 > > [3254102] > type=friend > ... > mohsuggest=class2 > > A call is bridged between the two extensions. When 3254102 > puts 3254101 on hold, 3254101 hears moh class 'class2' which > is correct. However, when 3254101 puts 3254102 on hold, the > 3254102 hears the default music class. > > Why? > > Doug. > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Running Multiple Instances of Asterisk
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes: DG> I'd like to know if anyone has sucessfully managed to run multiple DG> instances of Asterisk on the same system. - Did you run each DG> instance as a separate user? - Did you have any install or config DG> problems? - It looks like the G729 codec registration utility DG> doesn't work when files aren't installed in standard places. Did DG> you have this problem? - How many instances could be run on a DG> single Asterisk box? We run multiple asterisks with vserver. It has a slight disk space and memory penalty, but nothing compared to proper paravirtualisation or true virtualisation. It works very well. They are SIP only though -- it would be a bit more difficult if there was hardware involved. Most people on this list think one asterisk instance will cut it for tens or even hundreds of business customers. Best of luck to them, of course. I guess the first thing they do is replace callgroups and pickupgroups with something else. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with zaptel 1.4b2 and X101P Wildcard
On 9/26/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
Next thing to do, I guess, is to run:strace ztcfgto see which device exactly is accessed . Though /dev/zap/ctl is the
usual suspect.
[EMAIL PROTECTED] zaptel-1.4.0-beta1]# strace ztcfgexecve("/sbin/ztcfg", ["ztcfg"], [/* 25 vars */]) = 0uname({sys="Linux", node="mythtv", ...}) = 0brk(0) = 0x8779000
access("/etc/ld.so.preload", R_OK) = -1 ENOENT (No such file or directory)open("/etc/ld.so.cache", O_RDONLY) = 3fstat64(3, {st_mode=S_IFREG|0644, st_size=87164, ...}) = 0old_mmap(NULL, 87164, PROT_READ, MAP_PRIVATE, 3, 0) = 0xf6fea000
close(3) = 0open("/lib/tls/libm.so.6", O_RDONLY) = 3read(3, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\0\363C"..., 512) = 512fstat64(3, {st_mode=S_IFREG|0755, st_size=215248, ...}) = 0
old_mmap(0x43c000, 139424, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x43c000old_mmap(0x45d000, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0x45d000close(3) = 0
open("/lib/tls/libc.so.6", O_RDONLY) = 3read(3, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0 \0372\000"..., 512) = 512fstat64(3, {st_mode=S_IFREG|0755, st_size=1512400, ...}) = 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xf6fe9000old_mmap(0x30d000, 1207532, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 3, 0) = 0x30d000old_mmap(0x42e000, 16384, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x12) = 0x42e000
old_mmap(0x432000, 7404, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x432000close(3) = 0old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xf6fe8000
mprotect(0x42e000, 8192, PROT_READ) = 0mprotect(0x45d000, 4096, PROT_READ) = 0mprotect(0x309000, 4096, PROT_READ) = 0set_thread_area({entry_number:-1 -> 6, base_addr:0xf6fe86c0, limit:1048575, seg_32bit:1, contents:0, read_exec_only:0, limit_in_pages:1, seg_not_present:0, useable:1}) = 0
munmap(0xf6fea000, 87164) = 0open("/dev/zap/ctl", O_RDWR) = 3brk(0) = 0x8779000brk(0x879a000) = 0x879a000open("/etc/zaptel.conf", O_RDONLY) = 4
fstat64(4, {st_mode=S_IFREG|0644, st_size=9532, ...}) = 0mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0xf6fff000read(4, "#\n# Zaptel Configuration File\n#\n"..., 4096) = 4096
read(4, "he list remains idle\n# \"clear\" "..., 4096) = 4096read(4, "le is a single tone DCS transmit"..., 4096) = 1340read(4, "", 4096) = 0close(4) = 0
munmap(0xf6fff000, 4096) = 0ioctl(3, 0x80844a05, 0xfef3aad0) = -1 EINVAL (Invalid argument)ioctl(3, 0x404c4a13, 0x807a7cc) = -1 ENOTTY (Inappropriate ioctl for device)write(2, "ZT_CHANCONFIG failed on channel "..., 71ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)
) = 71close(3) = 0exit_group(1) = ?
/Morten
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Yes, have multiple clients with asterisk behind a sonicwall. I don't understand from your wording if you mean a voip connection suddenly changed from dup/5060, or, did you change the asterisk system to use some other udp port. The sonicwall does have an option to support sip (udp/5060), but I've not had to use it on anything that we've worked with. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE406P not working on Intel D101Ggc motherboard.
I recently moved a TE406P card from an Intel D865GBF motherboard (where it worked fine), to an Intel D101Ggc card, and now I can't get the spans to got up correctly. All I get is an endless burst of: == Primary D-Channel on span 4 up == Primary D-Channel on span 2 up !! Got a UA, but i'm in state 1 The other 2 spans doesn't even make an attempt to start. If I set up one of the "semi-awake" spans in intense debug mode, all that I get is: < Supervisory frame: < SAPI: 00 C/R: 1 EA: 0 < TEI: 000EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 000 P/F: 1 < 0 bytes of data -- ACKing all packets from 0 to (but not including) 0 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Unsolicited RR with P/F bit, responding Sending Receiver Ready (0) > [ 02 01 01 01 ] > Supervisory frame: > SAPI: 00 C/R: 1 EA: 0 > TEI: 000EA: 1 > Zero: 0 S: 0 01: 1 [ RR (receive ready) ] > N(R): 000 P/F: 1 > 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter and so on... The spans are connected to Pika Primenet E1 card, which is set as Network side in the four spans. This card is installed in a different machine. My configuration for the Digium card is: * zaptel.conf: span=1,1,0,ccs,hdb3,yellow bchan=1-15 dchan=16 bchan=17-31 span=2,2,0,ccs,hdb3,yellow bchan=32-46 dchan=47 bchan=48-62 span=3,3,0,ccs,hdb3,yellow bchan=63-77 dchan=78 bchan=79-93 span=4,4,0,ccs,hdb3,yellow bchan=94-108 dchan=109 bchan=110-124 loadzone=cl defaultzone=cl * zapata.conf: [channels] language=es switchtype=euroisdn pridialplan=national prilocaldialplan=national priindication=inband usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes relaxdtmf=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=-3.0 txgain=-15.0 canpark=yes resetinterval=never signalling=pri_cpe group=1 context=zap_pstn channel => 1-15 channel => 17-46 channel => 48-77 channel => 79-108 channel => 110-124 * Zaptel version 1.2.9.1, Asterisk 1.2.12.1, RHEL4 2.6.9-42.0.2.ELsmp, /proc/interrupts says: CPU0 CPU1 0: 47653469 47665247IO-APIC-edge timer 1: 8 0IO-APIC-edge i8042 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 12: 67 0IO-APIC-edge i8042 14: 446945 446248IO-APIC-edge ide0 169: 0 0 IO-APIC-level libata 177: 30424 29392 IO-APIC-level libata 185: 0 0 IO-APIC-level ehci_hcd, ohci_hcd, ohci_hcd 193:222 0 IO-APIC-level HDA Intel 201: 47669502 47599256 IO-APIC-level wct4xxp 209: 129534 0 IO-APIC-level eth0 NMI: 0 0 LOC: 95329567 95329645 ERR: 0 MIS: 0 IRQ balance is running. Thanks a lot for your answers. -- Atly. Alvaro Palma. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Set hint status from dialplan?
Here is an output from a 1.4.0-Beta2 voipgw*CLI> show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- Agent Call Agent Proxy Channel yes yes no Console OSS Console Channel Driver no yes no Zap Zapata Telephony Driver w/PRIno yes no Skinny Skinny Client Control Protocol (Skinny) no yes no Phone Standard Linux Telephony API Driver no yes no Feature Feature Proxy Channel Driver no yes no SIP Session Initiation Protocol (SIP)yes yes yes Local Local Proxy Channel Driver yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes MGCPMedia Gateway Control Protocol (MGCP)yes yes no -- 10 channel drivers registered. voipgw*CLI> show version Asterisk 1.4.0-beta2 built by root @ voipgw on a i686 running Linux on 2006-09-25 00:49:44 UTC voipgw*CLI> -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, September 26, 2006 2:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: Set hint status from dialplan? On Tuesday 26 September 2006 13:57, C F wrote: > Andrew what does "show channeltypes" give you? *CLI> show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- Zap Zapata Telephony Driver w/PRIno yes no SIP Session Initiation Protocol (SIP)yes yes yes Local Local Proxy Channel Driver yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no -- 6 channel drivers registered. *CLI> show version Asterisk SVN-trunk-r41990 built by root @ asterisk on a i686 running Linux on 2006-09-12 03:02:05 UTC Curious... I see Local/ has a devicestate, and I've never heard of a "Feature/" channel type before... :-) So I imagine I could use Local/[EMAIL PROTECTED] to see parking slot state, but nothing for arbitrary channels such as what Lacy is showing. Is that correct? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemailmain menu
Hi, Is there way a way to restrict access to certain menus, such as the following: 0 Mailbox options 1 Record your unavailable message 2 Record your busy message 3 Record your name 4 Record your temporary message (new in Asterisk v1.2) Thanks in advance, Jack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk with cisco 7935
I would also recommend either the Polycom IP4000, or the Clearone MAXIP, both of which are SIP native. If cost is an issue, you can also take an inexpensive Polycom analog conference phone, such as the Voicestation 100, and SIP enable it using a Linksys SPA-1001 analog adapter. For about half the price of a native SIP conference phone you have a working solution. Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Amos Sent: Tuesday, September 26, 2006 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk with cisco 7935 I spent quite a bit of time debugging the 7935/7936, and it is an issue inside the firmware that Cisco knows how to work around in CallManager. There are better conference phone options available, and development on chan_sccp is basically dead at this point anyway, so I don’t see this one ever being fixed. I would recommend a Polycom IP4000, it’s the exact same phone body but is much cheaper MSRP, and it’s SIP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Tuesday, September 26, 2006 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with cisco 7935 Just wondering if anyone has had any luck getting the cisco 7935 working with asterisk and if so, what is the best way to go about it? on the The consensus on the chan_sccp list is that it seems to be a good door stop. Seems something is just different about its SCCP image. There is new SCCP firmware that was released this month. I don't know if it works any better. -- Lacy Moore I'm the guy that doesn't give a damn about anyone's problems but my own... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rewriting CID number w/o changing CDR src field
Hi all. As a convieneince to my users, I'm trying to strip off the leading 1 and areacode from incoming calls. However, when I do, the src field in the CDR is also stripped. I'd like the CDR to reflect the "connonical" form of the incoming number. Any way do to this? TIA, Mike Diehl. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Set hint status from dialplan?
On Tuesday 26 September 2006 13:57, C F wrote: > Andrew what does "show channeltypes" give you? *CLI> show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- Zap Zapata Telephony Driver w/PRIno yes no SIP Session Initiation Protocol (SIP)yes yes yes Local Local Proxy Channel Driver yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes Feature Feature Proxy Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no -- 6 channel drivers registered. *CLI> show version Asterisk SVN-trunk-r41990 built by root @ asterisk on a i686 running Linux on 2006-09-12 03:02:05 UTC Curious... I see Local/ has a devicestate, and I've never heard of a "Feature/" channel type before... :-) So I imagine I could use Local/[EMAIL PROTECTED] to see parking slot state, but nothing for arbitrary channels such as what Lacy is showing. Is that correct? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk with cisco 7935
I spent quite a bit of time debugging the 7935/7936, and it is an issue inside the firmware that Cisco knows how to work around in CallManager. There are better conference phone options available, and development on chan_sccp is basically dead at this point anyway, so I don’t see this one ever being fixed. I would recommend a Polycom IP4000, it’s the exact same phone body but is much cheaper MSRP, and it’s SIP. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Tuesday, September 26, 2006 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with cisco 7935 Just wondering if anyone has had any luck getting the cisco 7935 working with asterisk and if so, what is the best way to go about it? on the The consensus on the chan_sccp list is that it seems to be a good door stop. Seems something is just different about its SCCP image. There is new SCCP firmware that was released this month. I don't know if it works any better. -- Lacy Moore I'm the guy that doesn't give a damn about anyone's problems but my own... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H323 IP phones
Hi guys!Can someone give advice on nice H323 IP phones brands?? I'm looking for some H323 IP phones for a customer. Diving in theinternet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard about them. Can someone give feedback experince about it??, configease, sound quality, visual appearance, end-user feedback, any infowill be appreciated.thnx!Alyed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Thanks All I have those settings already enabled It is rejecting the SIP INVITE packet not even getting to Voice at all The VoIP provider shows a registered with a good Qualify time 55 ms but not calls come in due to the Invite packet being rejected Why and why would it suddenly do this nothing was changed it blink I'm not going to work now Thanks all Barry Dr. Michael J. Chudobiak wrote: Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Which Sonicwall model? Some (like the TZ170) have special VOIP settings, like "Enable consistent NAT" and "Enable SIP Transformations". Check those; they work well with SIP. If you don't have one of these newer models, please see http://www.voip-info.org/wiki-IAX, in the "NAT Issues" section. It deals with IAX2, but the issues are same for SIP UDP. The Sonicwall UDP-connection-memory timeout may be VERY short - 30 seconds by default on some! It is adjustable in some firmware versions. I use the TZ170, but with IAX2 rather than SIP. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Set hint status from dialplan?
Andrew what does "show channeltypes" give you? On 9/26/06, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: On Tuesday 26 September 2006 13:02, Steven wrote: > That is the metermaid patch. It has been included into 1.4 as far as I > know. I do not see "DevState" in my "show application" output, so I would say no, it's not in 1.4. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Included context
How about trying Noop(CONTEXT IS: ${CONTEXT}) before you post it? but
now that you have posted just try it and report back.
On 9/26/06, Artifex Maximus <[EMAIL PROTECTED]> wrote:
Hello,
For example I have this dialplan:
[context1]
exten => s,1,Noop
exten => s,n,Dial(...)
exten => s,n,Playback(${CONTEXT})
exten => s,n,Hangup
[context2]
include => context1
[context3]
include => context1
Then I make dial-out call files with context2, context3, etc. What is
the value of ${CONTEXT} in that case? Still context1 because it's
physically there or context2, context3 because I am included from
there (so in a way logically is there). I didn't find any exact answer
that's why I'm asking here.
bye,
Zsolt
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] PRI Outbound CallerID Question
There seems to be three tiers in my experience: 1. Only your DID's 2. Arbitrary, but the pilot number of the PRI will appear if you suppress your Caller ID 3. Completely arbitrary, including <--this is the fa shizzle So you want 2) or 3) but definitely it is a telco thing. You need to sweetly social engineer someone in the call centre at your telco. -Original Message-From: Lacy Moore - Aspendora [mailto:[EMAIL PROTECTED]Sent: Tuesday, September 26, 2006 11:50 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] PRI Outbound CallerID Question Yes, it is possible. But, your Telco has to support this. Your Telco has to give you the ability to set your caller ID. Some providers (and it sounds like yours may be one of them) only allow you to use numbers which you are authorized to use (such as your DIDs). On 9/26/06, Shawn Kelley <[EMAIL PROTECTED]> wrote: Hi all,I've searched around and haven't found much of an answer to my issue. Anyadvice from you would be appreciated. Problem: Need to take an inbound call from our PRI and forward it to anotherPSTN user via the PRI, sending the original callers id with it.I know this can be done since we currently use an 800 service that does it. You call the 800 number; they answer and put you on hold. They then outcallto the pstn numbers we have defined and the incoming call shows up with theoriginal callers CID, we answer and have options to accept or reject the call.So I know the 800 provider is staying in the middle of the call and not justperforming a redirect to us.I've tried the various CID settings in Asterisk, but am not able to useanything but our DID numbers for our outbound caller id. My telco has been unresponsive to this issue.Does anyone know if it's possible with a PRI or do you have to have someother type of PSTN connection such as SS7?Thanks!!--Shawn___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Set hint status from dialplan?
On Tuesday 26 September 2006 13:25, Lacy Moore - Aspendora wrote: > http://forums.digium.com/viewtopic.php?t=891&highlight=shared+line Direct link for those of us who can't stand forums: http://bugs.digium.com/view.php?id=5779 -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
Besides for what Lacy answered, have you tried NOT playing with setting CID? Just do a blind xfer, or just use dial whatever on the DID itself. If that doesn't work then like Lacy said your provider might be blocking it. On 9/26/06, Shawn Kelley <[EMAIL PROTECTED]> wrote: Hi all, I've searched around and haven't found much of an answer to my issue. Any advice from you would be appreciated. Problem: Need to take an inbound call from our PRI and forward it to another PSTN user via the PRI, sending the original callers id with it. I know this can be done since we currently use an 800 service that does it. You call the 800 number; they answer and put you on hold. They then outcall to the pstn numbers we have defined and the incoming call shows up with the original callers CID, we answer and have options to accept or reject the call. So I know the 800 provider is staying in the middle of the call and not just performing a redirect to us. I've tried the various CID settings in Asterisk, but am not able to use anything but our DID numbers for our outbound caller id. My telco has been unresponsive to this issue. Does anyone know if it's possible with a PRI or do you have to have some other type of PSTN connection such as SS7? Thanks!! --Shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Gateway
I am thinking of using a mini atx 1u server with a digium zaptel (wcte11xp) installed to act as a SIP gateway. This way any of my asterisk servers can forward calls to any gateway (seperated by about 3miles of fiber). Has anyone else tried this? I would just load a basic asteisk config and zaptel with something like CentOS 4.4 ServerCD. Here is the hardware I am thinking of. http://www.abmx.com/1u-short-depth-rack-mount-server-p-256.html It seems like this would be alot cheaper than getting a pre-built sip gateway from VOX. Any input is greatly appreciated. Forrest ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Outbound CallerID Question
Yes, it is possible. But, your Telco has to support this. Your Telco has to give you the ability to set your caller ID. Some providers (and it sounds like yours may be one of them) only allow you to use numbers which you are authorized to use (such as your DIDs). On 9/26/06, Shawn Kelley <[EMAIL PROTECTED]> wrote: Hi all,I've searched around and haven't found much of an answer to my issue. Anyadvice from you would be appreciated. Problem: Need to take an inbound call from our PRI and forward it to anotherPSTN user via the PRI, sending the original callers id with it.I know this can be done since we currently use an 800 service that does it. You call the 800 number; they answer and put you on hold. They then outcallto the pstn numbers we have defined and the incoming call shows up with theoriginal callers CID, we answer and have options to accept or reject the call.So I know the 800 provider is staying in the middle of the call and not justperforming a redirect to us.I've tried the various CID settings in Asterisk, but am not able to useanything but our DID numbers for our outbound caller id. My telco has been unresponsive to this issue.Does anyone know if it's possible with a PRI or do you have to have someother type of PSTN connection such as SS7?Thanks!!--Shawn ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Included context
Hello,
For example I have this dialplan:
[context1]
exten => s,1,Noop
exten => s,n,Dial(...)
exten => s,n,Playback(${CONTEXT})
exten => s,n,Hangup
[context2]
include => context1
[context3]
include => context1
Then I make dial-out call files with context2, context3, etc. What is
the value of ${CONTEXT} in that case? Still context1 because it's
physically there or context2, context3 because I am included from
there (so in a way logically is there). I didn't find any exact answer
that's why I'm asking here.
bye,
Zsolt
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Set hint status from dialplan?
Steven, If you are trying to do this on a stock Asterisk system (and I can certainly understand why you would want to), then what I have implemented will definitely not work. I couldn't find anyway to do this on a stock system. Upgrades are going to be a nightmare with all the patches that have been applied to my system. This was something that was absolutely needed, and the patch that started me down that road was something that was absolutely needed. I had to have a completely "dummy proof" way for someone to park a call and pick it up at another extension. When I say "dummy proof", I mean it, too :-) That patches mentioned in the forum link do that very well, and as an added bonus, took care of my night mode indicator. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Set hint status from dialplan?
;exten => 799,hint,DS/mmgc Lacy, What is the DS/mmgc? The DS is what the DevState patch adds. I actually got to this point by following this thread: http://forums.digium.com/viewtopic.php?t=891&highlight=shared+line After I implemented these changes, I had the DevState on the system. The mmgc is just an arbitrary name you can use. We have several companies sharing the same phone system, and that is one of the companies. The Polycom monitors the hint status of extension 799, which is the Device State of mmgc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI Outbound CallerID Question
Hi all, I've searched around and haven't found much of an answer to my issue. Any advice from you would be appreciated. Problem: Need to take an inbound call from our PRI and forward it to another PSTN user via the PRI, sending the original callers id with it. I know this can be done since we currently use an 800 service that does it. You call the 800 number; they answer and put you on hold. They then outcall to the pstn numbers we have defined and the incoming call shows up with the original callers CID, we answer and have options to accept or reject the call. So I know the 800 provider is staying in the middle of the call and not just performing a redirect to us. I've tried the various CID settings in Asterisk, but am not able to use anything but our DID numbers for our outbound caller id. My telco has been unresponsive to this issue. Does anyone know if it's possible with a PRI or do you have to have some other type of PSTN connection such as SS7? Thanks!! --Shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Set hint status from dialplan?
On Tuesday 26 September 2006 13:02, Steven wrote: > That is the metermaid patch. It has been included into 1.4 as far as I > know. I do not see "DevState" in my "show application" output, so I would say no, it's not in 1.4. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Set hint status from dialplan?
That is the metermaid patch. It has been included into 1.4 as far as I know.
I am hoping to use that for parking slot BLFs on the phones.
My extension for day/night mode is not a real channel, so I am hoping to set
the hint value manually.
--
--
Steven
http://www.glimasoutheast.org
"C F" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> IIRC, there was a dev status for the local channel being worked on the
> bug tracker.
> Ok, here is the link:
> http://bugs.digium.com/view.php?id=5779
>
> On 9/26/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:
>>
>>
>> > Is it possible to manually set the hint status of a virtual extension via
>> the dialplan?
>> >
>> > I have an extension that turns my night mode on and off.
>> >
>> > I would love to be able to manually set the hint to be able to turn on a
>> light for night mode.
>> >
>> >
>>
>> See
>> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate
>> for more info on this.
>>
>> Here is a part of my extensions.conf that uses this:
>>
>> ; Night Mode Activations
>> exten => 799,hint,DS/mmgc
>> exten =>
>> 799,1,Macro(open-close,mmgc,custom/moore-moore-now-open,custom/moore-moore-now-closed)
>> exten => 799,n,Playback(beep)
>> exten => 799,n,Hangup
>>
>>
>>
>> [macro-open-close]
>> exten => s,1,DBGet(nightmode=nightmode/${ARG1})
>> exten => s,n,NoOp(ARG1 ${ARG1} ARG2 ${ARG2} ARG3 ${ARG3})
>> exten => s,n,Set(OpenFile=${ARG2})
>> exten => s,n,Set(CloseFile=${ARG3})
>> exten => s,n,NoOp(Close file ${CloseFile}. Open file ${OpenFile}.)
>> exten => s,n,GotoIf(${nightmode}=1?s,Open:s,Close)
>> exten => s,n(Open),DBPut(nightmode/${ARG1}=0)
>> exten => s,n,Devstate(${ARG1},0)
>> exten => s,n,Playback(${OpenFile})
>> exten => s,n,Goto(Return)
>> exten => s,n(Close),DBPut(nightmode/${ARG1}=1)
>> exten => s,n,Devstate(${ARG1},2)
>> exten => s,n,Playback(${CloseFile})
>> exten => s,n(Return),NoOp
>>
>> On my Polycom, I have a speeddial set up for 799. One press, and it turns
>> night mode on and announces that, another press and it turns night mode off
>> and announces that..
>>
>>
>> --
>> Lacy Moore
>> I'm the guy that doesn't give a damn about anyone's problems but my own...
>> ___
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Which Sonicwall model? Some (like the TZ170) have special VOIP settings, like "Enable consistent NAT" and "Enable SIP Transformations". Check those; they work well with SIP. If you don't have one of these newer models, please see http://www.voip-info.org/wiki-IAX, in the "NAT Issues" section. It deals with IAX2, but the issues are same for SIP UDP. The Sonicwall UDP-connection-memory timeout may be VERY short - 30 seconds by default on some! It is adjustable in some firmware versions. I use the TZ170, but with IAX2 rather than SIP. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Set hint status from dialplan?
;exten => 799,hint,DS/mmgc
Lacy, What is the DS/mmgc?
-- -- Steven
http://www.glimasoutheast.org
"Lacy Moore - Aspendora" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]...
Is
it possible to manually set the hint status of a virtual extension via the
dialplan?I have an extension that turns my night mode on and off.
I would love to be able to manually set the hint to be able to turn
on a light for night mode.
See http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate for
more info on this.
Here is a part of my extensions.conf that uses this:
; Night Mode Activationsexten => 799,hint,DS/mmgcexten =>
799,1,Macro(open-close,mmgc,custom/moore-moore-now-open,custom/moore-moore-now-closed)exten
=> 799,n,Playback(beep)exten => 799,n,Hangup
[macro-open-close]exten =>
s,1,DBGet(nightmode=nightmode/${ARG1})exten => s,n,NoOp(ARG1 ${ARG1}
ARG2 ${ARG2} ARG3 ${ARG3})exten => s,n,Set(OpenFile=${ARG2})exten
=> s,n,Set(CloseFile=${ARG3}) exten => s,n,NoOp(Close file
${CloseFile}. Open file ${OpenFile}.)exten =>
s,n,GotoIf(${nightmode}=1?s,Open:s,Close)exten =>
s,n(Open),DBPut(nightmode/${ARG1}=0)exten => s,n,Devstate(${ARG1},0)
exten => s,n,Playback(${OpenFile})exten =>
s,n,Goto(Return)exten => s,n(Close),DBPut(nightmode/${ARG1}=1)exten
=> s,n,Devstate(${ARG1},2)exten =>
s,n,Playback(${CloseFile})exten => s,n(Return),NoOp
On my Polycom, I have a speeddial set up for 799. One press, and it
turns night mode on and announces that, another press and it turns night mode
off and announces that..
-- Lacy MooreI'm the guy that doesn't give a
damn about anyone's problems but my own...
___--Bandwidth and
Colocation provided by Easynews.com --asterisk-users mailing
listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Thanks all ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.lassologic.com/support/pdfs/Configuring_Voip_For_SonicOS_Enhanced.pdf#search=%22sonicos%20voip%22 -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk behind Sonicwall firewall
Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? SIP is still on 5060, but the AUDIO (which is RTP) is on a dynamically negotiated port. Now you understand why many people in the VoIP business would love to meet the people that designed SIP in a dark alley. Read the mailing list archives and the Wiki for information working around these issues. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk behind Sonicwall firewall
Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Thanks all ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk to cell phone network
Ciao Michiel, > > > http://www.junghanns.net/en/GSM-PCI_produkt.html > > > > > > If they are as stable as the quad/octo BRI cards they have > > > it's a real winner. > > > > Where can I see the prices of this cards? > > My supplier has them listed as: > UnoGSM: 900 euro > DuoGSM: 1200 euro > QuadGSM: 1600 euro Well, how does Asterisk interact with those devices? Is there a chan_gsm_pci? Thanks, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] X100P Clone card in JAPAN
Did anyone success to install X100P Clone card on Asterisk to work with Japan stanadards for Analog line over ISDN TA. I can't make call when the call is ringing is OK and in the moment when the call is pick up the line is droped. I have hang ups all the time. I can't make call. Did anyone else have this problem. Is it depends on the Japanes provider or ? Help Please. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk to cell phone network
On 10:25, Tue 26 Sep 06, Tomislav Par?ina wrote: > In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... > > 1/2/4 simslot pci card: > > http://www.junghanns.net/en/GSM-PCI_produkt.html > > > > If they are as stable as the quad/octo BRI cards they have > > it's a real winner. > > Where can I see the prices of this cards? My supplier has them listed as: UnoGSM: 900 euro DuoGSM: 1200 euro QuadGSM: 1600 euro -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD "Why is it drug addicts and computer afficionados are both called users?" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set hint status from dialplan?
IIRC, there was a dev status for the local channel being worked on the
bug tracker.
Ok, here is the link:
http://bugs.digium.com/view.php?id=5779
On 9/26/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:
> Is it possible to manually set the hint status of a virtual extension via
the dialplan?
>
> I have an extension that turns my night mode on and off.
>
> I would love to be able to manually set the hint to be able to turn on a
light for night mode.
>
>
See
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+BristuffDevstate
for more info on this.
Here is a part of my extensions.conf that uses this:
; Night Mode Activations
exten => 799,hint,DS/mmgc
exten =>
799,1,Macro(open-close,mmgc,custom/moore-moore-now-open,custom/moore-moore-now-closed)
exten => 799,n,Playback(beep)
exten => 799,n,Hangup
[macro-open-close]
exten => s,1,DBGet(nightmode=nightmode/${ARG1})
exten => s,n,NoOp(ARG1 ${ARG1} ARG2 ${ARG2} ARG3 ${ARG3})
exten => s,n,Set(OpenFile=${ARG2})
exten => s,n,Set(CloseFile=${ARG3})
exten => s,n,NoOp(Close file ${CloseFile}. Open file ${OpenFile}.)
exten => s,n,GotoIf(${nightmode}=1?s,Open:s,Close)
exten => s,n(Open),DBPut(nightmode/${ARG1}=0)
exten => s,n,Devstate(${ARG1},0)
exten => s,n,Playback(${OpenFile})
exten => s,n,Goto(Return)
exten => s,n(Close),DBPut(nightmode/${ARG1}=1)
exten => s,n,Devstate(${ARG1},2)
exten => s,n,Playback(${CloseFile})
exten => s,n(Return),NoOp
On my Polycom, I have a speeddial set up for 799. One press, and it turns
night mode on and announces that, another press and it turns night mode off
and announces that..
--
Lacy Moore
I'm the guy that doesn't give a damn about anyone's problems but my own...
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Multiple Instances of Asterisk
On Tue, Sep 26, 2006 at 06:03:46PM +0300, Tzafrir Cohen wrote: > On Mon, Sep 25, 2006 at 08:58:52PM +0100, Julian Lyndon-Smith wrote: > > Jay R. Ashworth wrote: > > > voicemail.conf doesn't, as it needs to be modified by app_voicemail for > > password changes. > > An alternative is to use an external script to modify that file. Careful with the quoting there, Tzafrir: (it correctly indicates) that I didn't actually say that, and I'm not the person who cares, anyway. :-) That said, thanks for at least *clipping the quotes* in much the same way that almost everyone else doesn't. Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 "That's women for you; you divorce them, and 10 years later, they stop having sex with you." -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: asterisk t.38 (was RE: [asterisk-users] trixbox t38 pass through)
marek cervenka wrote: T38 passthrough doesn't seem to work in trunk at the moment. that's true http://bugs.digium.com/view.php?id=7679 http://bugs.digium.com/view.php?id=7844 t.38 in asterisk 1.4 http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 I've taken the code in Openpbx somewhat farther than the code in Asterisk SVN. Openpbx is now working for a lot of T.38 passthrough scenarios, and T.38 termination is now fairly solid. T.38 gateway is also basically working, though I haven't yet handed that out to anyone else for further testing. The big thing that had to change was to reuse the RTP port for the UDPTL stream. The code I donated to * was based on the specs. We found too many things that just don't work if you simply follow the specs. To make the software tolerant of a lot of other boxes doing weird things it seems you really have to reuse the RTP port for the UDPTL stream. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] University dumps CISCO VoIP for Asterisk
On Wednesday 20 September 2006 21:40, Douglas Garstang wrote: > We stuck OpenSER in between the phones and Asterisk, and pointed our phones > towards the OpenSER boxes for SIP registrations and subscriptions. When > OpenSER received a REGISTER or SUBSCRIBE message, it would use the send() > command to forward the messages onto each Asterisk server. By doing that, > ALL of our Asterisk servers had a copy of all sip registrations and > subscriptions. It seemed to work pretty well, but for unrelated reasons, we > dropped that approach. Which approach do you use now? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Dual core
Here's what we set in menuconfig when building Linux kernels for multi-processor systems: Processor Type and Features ---> ->Symmetric multi-processing support ->Timer frequency (1000 HZ) Device Drivers ---> Character devices ---> <*> Enhanced Real Time Clock Support Real Time Clock ---> <*> RTC class MATT--- On 9/26/06, Raphaël Jacquot <[EMAIL PROTECTED]> wrote: Matt Florell wrote: > For the Asterisk installation, no. For Linux, yes. I built a custom > SMP kernel, which depending on your Linux distribution may or may not > be necessary for you. > what specific things have you done, that isn't in the base kernel ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - alcatel
I'm trying the same in Alcatel 4200 and i solved changing ignaling from pri_cpe for pri_net.-- Frederico Madeira[EMAIL PROTECTED] 2006/9/26, Sylvain ZUCCA <[EMAIL PROTECTED]>: Hi, can you send logs from alcatel 4400 ? just log in with account mtcl and launch "t3" to see traces from the PBX Best Regards. 2006/9/26, et pourquoi pas ? epp <[EMAIL PROTECTED]>: Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). But I think there is a link with the fact that the digium card (110) is always yellow Do you have a idea for me ? Best regards, Thomas ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I stop lost DNS from killing Asterisk?
Rich Adamson wrote: Eric "ManxPower" Wieling wrote: Use IP addresses instead of hostnames in your Asterisk config. It sucks, but that is the only way I know of. Eric Bishop wrote: When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? Using IP addresses only does not fix the problem as the asterisk system does not know who he is. Need to define him in /etc/hosts as well, then it works just fine. A correctly set up system would already have that info in /etc/hosts, but it is a good thing to check because most systems are not correctly set up. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Play wav file during conversation
I want to be able to playback a certain soundfile for all parties in a call to hear. How would I do that? Eric ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Line Pickup Problem
Pato Valarezo wrote: Lacy Moore - Aspendora wrote: Wherever you have your exten => s,1,Answer statement, replace with: exten => s,1,Wait(30) ; or however long you want to wait to give someone else the chance to answer exten => s,n,Answer then continue on. Asterisk will then wait 30 seconds before it answers the phone. You would probably want this a lower number, though. Hi, i'm using x100P clones and i have two related issues: 1. In the first system (or in both) when someone answer the call, asterisk doesn't notice the stop ringing signal and continues with the dialplan, and of course answer the call and plays the welcome message and interrupts the current call in progress. 2. One of the system wich is connected to the PSTN doesn't seems to wait the time i specify in exten => s,1,Wait(10), and answers the line in a shorter time... it seems like the time doesn't count to it. I'm testing and training with this systems until i can buy a better quality hardware i expect to not have this problems with digium or better hardware. If someone has experience in this i'll apreciate comments. Based only on the words that you've used above, it sounds like you have a problem with extensions.conf (and maybe with the 'context' associated with the x100p card. To better understand your issue, we'll need to see your extensions.conf file and zapata.conf file contents. I'd suggest not trying to copy/paste a piece of those two files but rather include the entire files. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Opinions on Aastra 480i CT?
Hang onI have a 480i on my desk. The deskset definitely has a hold key. The programmable keys make a VM key really easy too. The cordless handset is limited by the number of buttons, but there are keystrokes for hold and a number of other functions. I wouldn't say that the cordless could be your main phone, but it certainly suuports the deskset well. Michael On Mon, 25 Sep 2006 09:31:42 -0600, Colin Anderson wrote: >>It's excellent home phone. I wouldn't use it in a business environment. >No >>hold, no one-touch voicemail. However, it works great! > >aw crap, that's a biggie but I think I can work around it, teach the user to >dial *98 for voicemail, *700 for park and hash to transfer, currently the >users dial feature-9-8-1 for voicemail right now so they are used to doing >things the hard way. But a dedicated hold and transfer button would've been >nice. The users' big requirement is inbound /outbound / missed call logging, >how is that? >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there T.38 support on asterisk 1.4 beta2 ???
Do anybody knows? Rgds, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztcfg / X100P question
On Tue, Sep 26, 2006 at 08:45:49AM -0400, Michel Vaillancourt wrote: > Tzafrir Cohen wrote: > > > > what's the contents of /etc/zaptel.conf ? > > > pbx1:~# cat /etc/zaptel.conf > # > # Zaptel Configuration File > # This file is parsed by the Zaptel Configurator, ztcfg > # > loadzone = us > defaultzone=us > # > fxsks=1 > # > channels=1 This line is unnecessary. Just remove it. > > > > > > This error comes from ztcfg . Look at ztcfg.c at the tarball of zaptel. > > Though from the word "tones" I gather that this is related to the > > tonezone library. > > > Unfortunately I am not a C programmer, and thus the file in > question is largely shrapnel to me. However, from what I can glean, > the function in question is "static int rad_chanconfig(char *keyword, > char *args)" and has something to do with "struct zt_radio_param". I > am puzzled as to what is going on that it thinks a radio is involved. The keyword "channels" you used is for some radio-related stuff. You configured zaptel.conf and zapata.conf... -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Multiple Instances of Asterisk
On Mon, Sep 25, 2006 at 08:58:52PM +0100, Julian Lyndon-Smith wrote: > Jay R. Ashworth wrote: > voicemail.conf doesn't, as it needs to be modified by app_voicemail for > password changes. An alternative is to use an external script to modify that file. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Errors
1 - Eclipse situation - What is inside fastagi-mapping.properties ? Are you using the sample HelloAgiScript from asterisk-java ? 2 - Command line situation - what's the command line you are using ? []'s, Edmilson Santana Unitech Tecnologia de Informação (http://www.unitech.com.br/) [EMAIL PROTECTED] wrote: i have all files in the same directory: c:\agi (asterisk-java-0.2.jar, fastagi-mapping.properties, HelloAgiScript.class and HelloAgiScript.java). My slasspath is also c:\agi Did you mean this? But i get still the following errors: if i start it with eclipse: ... INFO: Received connection. 25.09.2006 16:36:30 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: Unable to create AGIScript instance of type HelloAgiScript 25.09.2006 16:36:30 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: No script configured for URL 'agi://localhost.ch/hello.agi' (script 'hello.agi') if i start from the console another error occurs: INFO: Received connection. 25.09.2006 16:42:15 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: Resource bundle 'fastagi-mapping' is missing. 25.09.2006 16:42:15 net.sf.asterisk.util.impl.JavaLoggingLog error Severe: No script configured for URL 'agi://localhost/hello.agi' (scri pt 'hello.agi') What could that be? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk - alcatel
Hi, can you send logs from alcatel 4400 ? just log in with account mtcl and launch "t3" to see traces from the PBX Best Regards. 2006/9/26, et pourquoi pas ? epp <[EMAIL PROTECTED]>: Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 4400 and I use this: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI, but randomly, after few calls Astersik go in congestion (34). But I think there is a link with the fact that the digium card (110) is always yellow Do you have a idea for me ? Best regards, Thomas ___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sylvain ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I stop lost DNS from killing Asterisk?
Eric Bishop wrote: Hi All, When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? We just went through the same problem. You need both a caching dns server, and, define your asterisk system in /etc/hosts so he knows who he is. I've tested this several times as we use a laptop to demo asterisk and several of these demo's don't have any internet access. (And, you're right, asterisk does not process any calls.) With dns caching and the /etc/hosts definition in place, it now works everywhere. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: "does /var/run/asterisk.ctl exist?" -- butAsterisk *is* running.
I had a problem on one box where /var/run/asterisk/ did exist and had the correct non-root permissions. There was a typo in /etc/asterisk/asterisk.conf. was: astrundir => /var/run changed to: astrundir => /var/run/asterisk I do not remember which version of asterisk this was or if it was broken when adding freepbx. (or maybe when I tested [EMAIL PROTECTED]) I hope this helps. -- -- Steven http://www.glimasoutheast.org "Lacy Moore - Aspendora" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]... Be sure that it is looking in the right place. If it is running as non root, then the ctl file would be in a different directory. It looks as though Trixbox does run as non-root. The ctl is actually /var/run/asterisk/asterisk.ctl. Did you install from scratch, or was a previous version of Asterisk on the box? On 9/25/06, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote: Sorry if I'm stating the obvious, but I'm not sure if Trixbox runsasterisk as root or not. I have to "sudo asterisk -r" on mine, but I'm not running Trixbox, I'm running Asterisk 1.2.MojKen D'Ambrosio wrote:> I've set up a bunch of plain-jane Asterisk systems, but had heard good> things about the more recent incarnations of [EMAIL PROTECTED] errr, Trixbox.> So I installed it, and fired it up, and it works fine.>> Until I try to do an "asterisk -r". I get the "does /var/run/asterisk.ctl> exist?" question, which had always previously meant (to me) that Asterisk > wasn't running. But it is! And there's now asterisk.ctl file in the> entire /var hierarchy. Anyone have any ideas as to why that might be MIA?> It's insanely annoying, not being able to fire up the console. >> Thanks,>> -Ken>> ___> --Bandwidth and Colocation provided by Easynews.com -->> asterisk-users mailing list > To UNSUBSCRIBE or update options visit:>http://lists.digium.com/mailman/listinfo/asterisk-users>> !DSPAM:500,451866a1139541174510073! >--Mojo <[EMAIL PROTECTED]>Office Manager, Horan & Company, LLC(907) 747- x112___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Setting QOS settings in asterisk and/or CentOS?
Steven wrote: I found this command if your Cisco switches support it: "auto qos voip trust" You set this on each interface. It automatically prioritizes all SIP and skinny traffic, but not iax. There is also "auto qos voip cisco-phone". This one can detect a Cisco phone and prioritize it. I just have to figure out how to verify that it is actually doing anything. The auto qos function is a relatively new addition to the cisco routers and switches (eg, last year or so). The parameter is added to an individual interface (usually a serial interface), and it truly watches for actual traffic on that interface until you shut it down. At that point, auto qos "writes" the policy statements into the router config needed to support that actual traffic. To use it, you must: - enable it on an individual interface, - do not change the interface "bandwidth" statement while its running, - cisco express forwarding must be enabled, and, - all previously attached QoS policies must be removed from the interface being sampled. Its my understanding (although I've not actually done this) that auto qos can be used to monitor all traffic and not just voip packets. For example, some companies may wish to generate qos policies for Citrix, MS Terminal Server traffic, etc, and may not have any voip implementation at all. So, auto qos is not just for voip traffic and should be very usable with iax. Since you've specifically mentioned the "auto qos voip cisco-phone" statement, that statement essentially says watch for voip traffic coming from a cisco phone. Reading between the lines says: Cisco ships their voip phones with QoS already preconfigured with signaling traffic in one DSCP class and rtp traffic in another DSCP class. If your non-cisco phones aren't set up with those exact same DSCP markings, auto qos won't write the policy statements into your router's config. (E.g., cisco tends to push their proprietary voip sutff, so guess what... "auto qos voip cisco-phone" was oriented around those phones and not necessarily the sip versions of that same cisco phone.) The simplest command is "auto qos" applied to an individual interface without any other qualifying parameters. Keep in mind that auto qos is actually monitoring your traffic in real time, which assumes you've got voip phones, asterisk box, etc, already preconfigured to mark packets with TOS or DSCP bits. If that's not the case, then your voip traffic appears as "default" non-qos traffic and no policy will be written to the router's config. For testing purposes, auto qos can be applied to an interface then multiple voip test calls can be initiated manually. It would then write the appropriate policy statements into your config based on those voip test calls. In a large production world, one would apply auto qos to an interface and let it be for some much longer period of time (eg, hours). Then auto qos would write the config statements necessary to support the actual traffic observed over that period of time. There is no magic behind using auto qos; you can do the exact same thing manually by configuring policies in the router and doing something like "show policy-map interface s1". That display will tell you how much bandwidth is consumed for each QoS class that has been configured in your policy. The problem with doing that manually is that you have to know when your peak traffic period is for voip traffic, and then run the commands during that peak period to get it right. There are technical white papers on the cisco web site (somewhere) that describes how to use the auto qos function, but keep in mind the function was only recently introduced so it is not yet implemented on every product or in every IOS image. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I stop lost DNS from killing Asterisk?
Eric "ManxPower" Wieling wrote: Use IP addresses instead of hostnames in your Asterisk config. It sucks, but that is the only way I know of. Eric Bishop wrote: When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? Using IP addresses only does not fix the problem as the asterisk system does not know who he is. Need to define him in /etc/hosts as well, then it works just fine. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can I stop lost DNS from killing Asterisk?
Hi All,When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
