[Asterisk-Users] Fax with SPA-2000's?
I've been trying to get fax reception to work using an SPA-2000 to ring the fax machine or modem that's taking fax calls. I was curious if anyone else had tried something similar, and if so, had any luck getting it to work reliably. I've been able to get it to work, but it isn't reliable. (Pages/lines of black dots result more frequently than not.) The incoming lines are FXO and going to something digital isn't an option. My setup looks like this: POTS --- X100P FXO -- Asterisk -- SPA-2000 SIP -- Fax Would I be better off using spandsp and have * take the fax call? Would a PCI FXS card be the best solution instead of the SPA-2000? I still need to be able to send using a traditional fax machine, which I'm guessing will suffer the same problem with the SPA-2000 that reception has. If someone has been able to use the SPA-2000 to receive faxes reliably, I'd appreciate any tips or configuration settings. Here's my zapata settings: signalling=fxs_ks usecallerid=no echotraining=yes echocancel=yes echocancelwhenbridged=yes faxdetect=none context=inbound-analog1 channel = 1 faxdetect=incoming context=inbound-analog2 channel = 2 SPA-2000 is set to use ulaw only, changed Echo Supp Enable to No from default. -- Seth et lux in tenebris lucet Mattinen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WaitExten substitute
On Sat, 2004-06-19 at 14:10, Randy Bush wrote: i am using the freebsd port, which seems to not yet have WaitExten(), which i kinda want to use thusly [ext-666] exten = _.,1,SetVar(areacode=666) exten = _.,2,Background(zz-in-who) ; give them list of extns exten = _.,3,WaitExten(10) ; let them enter extn to call include = extensions include = applications include = speeddials exten = i,1,HangUp exten = t,1,HangUp I never heard of the app WaitExten but you could do the following: exten = _.,1,DigitTimeout,5 exten = _.,2,ResponseTimeout,10 exten = _.,3,SetVar(areacode=666) exten = _.,4,Background(zz-in-who) include = extensions include = applications include = speeddials exten = i,1,PlayBack(Invalid-Ext) exten = i,2,HangUp exten = t,1,PlayBack(NoExtensionEntered) exten = t,2,Hangup how do i hack this? Don't, I suggest you read the handbook, the wiki, and/or the archives. Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Big problem with Flash
Hi, i have connected my asterisk to an PBX (via cheap FXO-card). I must dial 7 for my door system and than again Flash-7 to open the door. I can repeat the Flash-7 as often as I want to open the door. my problem is to build the right extensions. exten = 777,1,Answer() exten = 777,2,Dial(ZAP/1/7,60,tTHg) exten = 777,3,Flash(ZAP/1) exten = 777,4,SendDTMF,7 exten = 777,5,Answer() exten = 777,5,Wait,10 I can dial 777 on my Asterisk-Phone (SIP) and I will be connected to the door system of the PBX. but I cant dial Flash-7 for open the door any hints? thank you thorsten gehrig
Re: [Asterisk-Users] Big problem with Flash
small point. * will not go to priority 3 unless Zap/1/7 hangs up You can send DTMF directly by the dial command by using the option D(digits) but am not sure about flash - Original Message - From: Thorsten Gehrig To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 1:04 PM Subject: [Asterisk-Users] Big problem with Flash Hi, i have connected my asterisk to an PBX (via cheap FXO-card). I must dial 7 for my door system - and than again Flash-7 to open the door. I can repeat the Flash-7 as often as I want to open the door. my problem is to build the right extensions. exten = 777,1,Answer() exten = 777,2,Dial(ZAP/1/7,60,tTHg) exten = 777,3,Flash(ZAP/1) exten = 777,4,SendDTMF,7 exten = 777,5,Answer() exten = 777,5,Wait,10 I can dial 777 on my Asterisk-Phone (SIP) and I will be connected to the door system of the PBX. but I cant dial Flash-7 for open the door. any hints? thank you thorsten gehrig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Big problem with Flash
Hi i know - THATS MY PROBLEM. But: I must first call the 7 (like I do) and Talk to the person on the door. Only I I want to open I must den a flash and a second 7 to open the door. any hints? regards thorsten -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Navnit Chachan Gesendet: Samstag, 19. Juni 2004 10:01 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Big problem with Flash small point. * will not go to priority 3 unless Zap/1/7 hangs up You can send DTMF directly by the dial command by using the option D(digits) but am not sure about flash - Original Message - From: Thorsten Gehrig To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 1:04 PM Subject: [Asterisk-Users] Big problem with Flash Hi, i have connected my asterisk to an PBX (via cheap FXO-card). I must dial 7 for my door system - and than again Flash-7 to open the door. I can repeat the Flash-7 as often as I want to open the door. my problem is to build the right extensions. exten = 777,1,Answer() exten = 777,2,Dial(ZAP/1/7,60,tTHg) exten = 777,3,Flash(ZAP/1) exten = 777,4,SendDTMF,7 exten = 777,5,Answer() exten = 777,5,Wait,10 I can dial 777 on my Asterisk-Phone (SIP) and I will be connected to the door system of the PBX. but I cant dial Flash-7 for open the door. any hints? thank you thorsten gehrig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Big problem with Flash
Let asterisk dial 7 and connect you to the door phone. You can then do a flash+7 on your hard phone itself to open the door. Am i missing something here? - Original Message - From: Thorsten Gehrig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 1:42 PM Subject: AW: [Asterisk-Users] Big problem with Flash Hi i know - THATS MY PROBLEM. But: I must first call the 7 (like I do) and Talk to the person on the door. Only I I want to open I must den a flash and a second 7 to open the door. any hints? regards thorsten -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Navnit Chachan Gesendet: Samstag, 19. Juni 2004 10:01 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Big problem with Flash small point. * will not go to priority 3 unless Zap/1/7 hangs up You can send DTMF directly by the dial command by using the option D(digits) but am not sure about flash - Original Message - From: Thorsten Gehrig To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 1:04 PM Subject: [Asterisk-Users] Big problem with Flash Hi, i have connected my asterisk to an PBX (via cheap FXO-card). I must dial 7 for my door system - and than again Flash-7 to open the door. I can repeat the Flash-7 as often as I want to open the door. my problem is to build the right extensions. exten = 777,1,Answer() exten = 777,2,Dial(ZAP/1/7,60,tTHg) exten = 777,3,Flash(ZAP/1) exten = 777,4,SendDTMF,7 exten = 777,5,Answer() exten = 777,5,Wait,10 I can dial 777 on my Asterisk-Phone (SIP) and I will be connected to the door system of the PBX. but I cant dial Flash-7 for open the door. any hints? thank you thorsten gehrig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Testing UK emergency dialing and LCR.
Hi Kevin, KW By the way, it's useful to map 911 and 112 onto your 999 KW route for the benefit of foreigners who don't know any better. Your point about 112 is very useful but slightly misguided; although the UK has used 999, nationally since 1938, (the world's first single number access for emergency services) 112 was mandated for pan European use from 1992 onwards. 112 is *not* for foreigners who don't know any better, it's for everyone in the EU to learn so that when you are anywhere in the EU you stand a fighting chance of getting hold of emergency help at the first attempt. 999 will continue to run in parallel with 112 for many years to come but 112 should be taught to children and adults alike as the universal number for emergency services. Some UK Telcos also provided support for 911 for a little while but I believe that this was officially frowned upon; I'm not sure what the policy is now. W As another thing, what is the correct method when using least cost W routing... If you have a branch office that has no outside line W connectivity directly routing its calls over IP to HQ the other end of W the country when you dial 999 it gets handled by the local call center W to your HQ rather than the branch office. It became apparent, back in 1999, when I was part of a team providing consultancy to a UK Telco for VoIP VPN launch, that a POTS line would be required locally at each branch office for power-fail compliance and to ensure that the OACs (Operator Assistance Centres) did not get confused about which location the emergency call was originated from. We discussed spoofing the branch office CLI in network at an SS7 level but that idea was shelved as there would have to have been an associated POTS line entry in the OAC database in the first place. At that time Cisco CPE had no way of utilising the power-fail POTS lines so a red 'phone was provided for use on each floor of the branch offices that only had VoIP VPN telephony. HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kevin Walsh Sent: 19 June 2004 02:56 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Testing UK emergency dialing and LCR. Wayne [EMAIL PROTECTED] wrote: Just wondering how people test your emergency dialing in the UK. Obviously you need to dial the 999 for emergency services, but am a bit unsure if this would go down too well with the operator with a 'sorry just testing' call. (you do all /test/ your emergency dialing dont you!?:-) ) I tend to test by unplugging the phone line and dialling 999. You can watch the log and see that the call attempted to route to the POTS line. You can then dial a real POTS number and watch the same route succeed. The emergency services get very upset if you call them to test, unless you've arranged to do so in advance and have an allotted time slot. You're right though; you can't be absolutely sure that the 999 route will work until you test it with a real call. Just start a fire before you call. That'll probably work. :-) As another thing, what is the correct method when using least cost routing... If you have a branch office that has no outside line connectivity directly routing its calls over IP to HQ the other end of the country when you dial 999 it gets handled by the local call center to your HQ rather than the branch office. If you need emergency services access in your branch office then you should get a single line into that office. The emergency services tend to rush to the destination they know is correct for that phone number. By the way, it's useful to map 911 and 112 onto your 999 route for the benefit of foreigners who don't know any better. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing UK emergency dialing and LCR.
- Original Message - From: Storer, Darren [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 11:27 AM Subject: RE: [Asterisk-Users] Testing UK emergency dialing and LCR. It became apparent, back in 1999, when I was part of a team providing consultancy to a UK Telco for VoIP VPN launch, that a POTS line would be required locally at each branch office for power-fail compliance and to ensure that the OACs (Operator Assistance Centres) did not get confused about which location the emergency call was originated from. We discussed spoofing the branch office CLI in network at an SS7 level but that idea was shelved as there would have to have been an associated POTS line entry in the OAC database in the first place. At that time Cisco CPE had no way of utilising the power-fail POTS lines so a red 'phone was provided for use on each floor of the branch offices that only had VoIP VPN telephony. Here in Denmark (where I live) the 112 service also have regular 8 digit phone numbers to be used for this kind of scenarios. Normally your location and associated 112 service centre will be resolved by the SS7 network thus you only need to know about the 112 number, BUT, with the indroduction of our new small IP Telephony companies (5000 customers) they will (manually) map the appropriate 8 digit phone number to your 112 entry when you sign-up bypassing the SS7 logic and dial directly the emergency centre for your area. My bet is that this also applies for other telco's, the question is if they are willing to disclose the information or not. -- Soren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Thousands of contexts?
Hi, -Original Message- But the thought is correct, use a database to store the data and one context that does a lookup into the database and populates your callerid. It is a better way of doing things. You could even host it in the ast_db and then it shouldn't be too slow as you aren't spawning any outside apps. Here's another scenario I'm working with: I am using contexts per user to 'include' the numberranges they are allowed to dial. Any suggestions how to do that without a context per user ? Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Thousands of contexts?
Hi Sim :) you seem to be one of the top candidates as a ast_data power tester :) On Sat, 2004-06-19 at 12:14, Florian Overkamp wrote: Hi, -Original Message- But the thought is correct, use a database to store the data and one context that does a lookup into the database and populates your callerid. It is a better way of doing things. You could even host it in the ast_db and then it shouldn't be too slow as you aren't spawning any outside apps. Here's another scenario I'm working with: I am using contexts per user to 'include' the numberranges they are allowed to dial. Any suggestions how to do that without a context per user ? Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK install
Hi Tim, I have done a very similar migration to the one you are suggesting. To start off with, you could buy a TDM card with 4 modules to allow you to pick 4 analogue channels from the Argent system or one or two x100 cards and pass them to Asterisk. As the Argent kit is very configurable, you can configure the POTS lines to the US standard and have callerID and other features not supported on standard UK lines. This will allow you to start using the system in a production environment without the business suffering as you learn the system. Once you are happy the system operates as you expect, you can connect it directly with the PRI card and bypass the Argent system. You will find that Asterisk does work differently to the Argent kit and trying to emulate the behaviour may not be as easy as you expect. It would be interesting to find out if anyone has tried using the Argent POTS units on Asterisk or if this is possible. You could always use the Grandstream ATA 286 and use your handsets on these. The cheap SIP phones (in my experience) dont play well with standard headsets. Stuart. On Fri, 2004-06-18 at 16:36, Tim Guy wrote: Well I'm slowly learning my way around asterisk although as yet I haven't had the chance to actually hook the system up to an ISDN line. I am going to migrate from an Argent Office setup. My only problem is keeping costs down on the phones. The Argent system is running about 30 POTS phones. Can someone suggest the cheapest option? Should I get some kind of large scale FXS box or would the cost of doing that on a large scale work out the same as getting cheap SIP phones? I have a large number of POTS phones with headsets so I would have to take that into account if I replaced the phones with SIP's In an ideal world Id like to convert a number of POTS to soft phones but as always its persuading the users that they can operate in the same way. Our Telco is NTL offering us an ISDN 30 style package. I assume this is a E100P card requirement? Any suggestions for good UK reseller or shall I get it direct from Digium? Anyhow, as I say I'm getting more functionality out of Asterisk than I ever did with (personally thinking) a very confusing Argent setup. I just hope that I can make it financially viable to do the install Cheers Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- | OpusVL: IT solutions for business | http://www.opusvl.com | T: 08717 50 40 02 | F: 08717 50 40 03 | E: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk -rx not working well
We're trying to use asterisk -r -x sip show peers to monitor sip phone availability. Sometimes the command shows the correct output but 9 times out of 10 all that is returned is: Name/usernameHost Mask Port Status with no listing. Anyone else see this behaviour? Stephan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Thousands of contexts?
Hi Michael, -Original Message- Hi Sim :) you seem to be one of the top candidates as a ast_data power tester :) I'm game, tell me more :-) Here's another scenario I'm working with: I am using contexts per user to 'include' the numberranges they are allowed to dial. Any suggestions how to do that without a context per user ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy when not registered
If I try to dial a SIP extension which is not connected/registered with *, I end up getting a busy indication and the call goes through to the busy voicemail message. The extension is listed in the sip.conf, but it is not connected at the time. Shouldn't it go to the unavailable voicemail message? Simon Brown ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Thousands of contexts?
check #sboost and open your eyes :) all that does not work as promised yet is: I can't seem to get context includes right MWI is tricky and I don't know how to specify odecs correctly Besides that it works fantasticly, even in chaos DB setup where the extension from he table triggers a ODBCget app :)) On Sat, 2004-06-19 at 13:39, Florian Overkamp wrote: Hi Michael, -Original Message- Hi Sim :) you seem to be one of the top candidates as a ast_data power tester :) I'm game, tell me more :-) Here's another scenario I'm working with: I am using contexts per user to 'include' the numberranges they are allowed to dial. Any suggestions how to do that without a context per user ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] using asterisk as sip registrar is not working for me
Have you tried pinging the phone from the asterisk box to see if there is connectivity ? if so try doing a network capture to see if asterisk is recieving registeration packets at all ? Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of smadi Sent: 19 June 2004 00:53 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] using asterisk as sip registrar is not working for me hi; i have the following topolgy: asterisk box set with public ip address 1.2.3.4 i have a snom200 sip phone that resides on a subnet with 192.168.0.10 address which i know have worked previously with vocal now my sip.conf file looks as follows: == [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls [phone1] type=friend host=dynamic dtmfmode=inband mailbox=1000 ; Mailbox for message waiting indicator context=sip callerid=Me 2124 = the phone does not seem to register. I tried to include the 192.168.0.10 defaultip address and i tried putting the static ip address of the dhcp server which is the gateway to which is the snow phone connected. is there anything specific that i must run to get the phone working? thanks m. smadi zia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk -rx not working well
It's exiting before the output finishes printing, it's a known bug and a timing issue There's a patch I put that's a hack to put in a timeout for exit, you should be able to find it on bugs.digium.com Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephan Wik Sent: Saturday, June 19, 2004 1:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk -rx not working well We're trying to use asterisk -r -x sip show peers to monitor sip phone availability. Sometimes the command shows the correct output but 9 times out of 10 all that is returned is: Name/usernameHost Mask Port Status with no listing. Anyone else see this behaviour? Stephan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
[Asterisk-Users] chan_modem dialout
can a voice modem make an outbound call? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax with SPA-2000's?
We have it working reliably on calls coming in over a T1, I don't remember the exact settings we used, but I will send on Monday when I'm back in the office. Make sure that everything is 711Ulaw only and that your fax or modem operates no faster than 9600, data through Asterisk analog lines won't go any faster than 9600. MATT--- -Original Message- From: Seth Mattinen [mailto:[EMAIL PROTECTED] Sent: Saturday, June 19, 2004 2:16 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fax with SPA-2000's? I've been trying to get fax reception to work using an SPA-2000 to ring the fax machine or modem that's taking fax calls. I was curious if anyone else had tried something similar, and if so, had any luck getting it to work reliably. I've been able to get it to work, but it isn't reliable. (Pages/lines of black dots result more frequently than not.) The incoming lines are FXO and going to something digital isn't an option. My setup looks like this: POTS --- X100P FXO -- Asterisk -- SPA-2000 SIP -- Fax Would I be better off using spandsp and have * take the fax call? Would a PCI FXS card be the best solution instead of the SPA-2000? I still need to be able to send using a traditional fax machine, which I'm guessing will suffer the same problem with the SPA-2000 that reception has. If someone has been able to use the SPA-2000 to receive faxes reliably, I'd appreciate any tips or configuration settings. Here's my zapata settings: signalling=fxs_ks usecallerid=no echotraining=yes echocancel=yes echocancelwhenbridged=yes faxdetect=none context=inbound-analog1 channel = 1 faxdetect=incoming context=inbound-analog2 channel = 2 SPA-2000 is set to use ulaw only, changed Echo Supp Enable to No from default. -- Seth et lux in tenebris lucet Mattinen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] Big problem with Flash
hi a) the hard phone is a sip phone (without flash keys) b) if there where a hard phone - the asterisk must pass through the flash - because the phone is always connected to asterisk - and asterisk is connectet to the doorphone... or? regards thorsten -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Navnit Chachan Gesendet: Samstag, 19. Juni 2004 10:29 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Big problem with Flash Let asterisk dial 7 and connect you to the door phone. You can then do a flash+7 on your hard phone itself to open the door. Am i missing something here? - Original Message - From: Thorsten Gehrig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 1:42 PM Subject: AW: [Asterisk-Users] Big problem with Flash Hi i know - THATS MY PROBLEM. But: I must first call the 7 (like I do) and Talk to the person on the door. Only I I want to open I must den a flash and a second 7 to open the door. any hints? regards thorsten -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Navnit Chachan Gesendet: Samstag, 19. Juni 2004 10:01 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Big problem with Flash small point. * will not go to priority 3 unless Zap/1/7 hangs up You can send DTMF directly by the dial command by using the option D(digits) but am not sure about flash - Original Message - From: Thorsten Gehrig To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 1:04 PM Subject: [Asterisk-Users] Big problem with Flash Hi, i have connected my asterisk to an PBX (via cheap FXO-card). I must dial 7 for my door system - and than again Flash-7 to open the door. I can repeat the Flash-7 as often as I want to open the door. my problem is to build the right extensions. exten = 777,1,Answer() exten = 777,2,Dial(ZAP/1/7,60,tTHg) exten = 777,3,Flash(ZAP/1) exten = 777,4,SendDTMF,7 exten = 777,5,Answer() exten = 777,5,Wait,10 I can dial 777 on my Asterisk-Phone (SIP) and I will be connected to the door system of the PBX. but I cant dial Flash-7 for open the door. any hints? thank you thorsten gehrig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival and asterisk
extension.conf
exten = 555,1,Answer
exten = 555,2,Festival('good morning')
exten = 555,3,Wait(2)
exten = 555,4,Hangup
What's the problem I'm facing? Thanks in advance.
Remove the quote marks... should be like ...Festival(good morning)
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Re: [Asterisk-Users] Big problem with Flash
I checked with my installation, asterisk passes the flash through to the destination. I am not sure with the sip phone. - Original Message - From: Thorsten Gehrig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 6:34 PM Subject: AW: [Asterisk-Users] Big problem with Flash hi a) the hard phone is a sip phone (without flash keys) b) if there where a hard phone - the asterisk must pass through the flash - because the phone is always connected to asterisk - and asterisk is connectet to the doorphone... or? regards thorsten -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Navnit Chachan Gesendet: Samstag, 19. Juni 2004 10:29 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Big problem with Flash Let asterisk dial 7 and connect you to the door phone. You can then do a flash+7 on your hard phone itself to open the door. Am i missing something here? - Original Message - From: Thorsten Gehrig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 1:42 PM Subject: AW: [Asterisk-Users] Big problem with Flash Hi i know - THATS MY PROBLEM. But: I must first call the 7 (like I do) and Talk to the person on the door. Only I I want to open I must den a flash and a second 7 to open the door. any hints? regards thorsten -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Navnit Chachan Gesendet: Samstag, 19. Juni 2004 10:01 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] Big problem with Flash small point. * will not go to priority 3 unless Zap/1/7 hangs up You can send DTMF directly by the dial command by using the option D(digits) but am not sure about flash - Original Message - From: Thorsten Gehrig To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 1:04 PM Subject: [Asterisk-Users] Big problem with Flash Hi, i have connected my asterisk to an PBX (via cheap FXO-card). I must dial 7 for my door system - and than again Flash-7 to open the door. I can repeat the Flash-7 as often as I want to open the door. my problem is to build the right extensions. exten = 777,1,Answer() exten = 777,2,Dial(ZAP/1/7,60,tTHg) exten = 777,3,Flash(ZAP/1) exten = 777,4,SendDTMF,7 exten = 777,5,Answer() exten = 777,5,Wait,10 I can dial 777 on my Asterisk-Phone (SIP) and I will be connected to the door system of the PBX. but I cant dial Flash-7 for open the door. any hints? thank you thorsten gehrig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: WaitExten substitute
I never heard of the app WaitExten http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20WaitExten but you could do the following: exten = _.,1,DigitTimeout,5 exten = _.,2,ResponseTimeout,10 exten = _.,3,SetVar(areacode=666) exten = _.,4,Background(zz-in-who) include = extensions include = applications include = speeddials exten = i,1,PlayBack(Invalid-Ext) exten = i,2,HangUp exten = t,1,PlayBack(NoExtensionEntered) exten = t,2,Hangup yes, i could; and i did. problem is that it does not seem to work; hence my posting. it plays the background and then falls through to invalid on the first keypress. i suspect this may be a sipura config issue again causing a double invite; but i am not sure. how do i hack this? Don't, I suggest you read the handbook, the wiki, and/or the archives. been there. done that. but thanks for the pointers. [ btw, search function in wiki is not real great, to be polite; but that issue is not local to this wiki ] randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: WaitExten substitute
On Jun 19, 2004, at 6:47 AM, Randy Bush wrote: [ btw, search function in wiki is not real great, to be polite; but that issue is not local to this wiki ] Yeah. Google: site:voip-info.org waitexten Works much better. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] enum problems with the latest CVS
Hi, I just recompiled * with the latest CVS. I am using enum in my extensions to dial first over the internet, if applicable. Everything was working perfectly, but now after installing the latest CVS I am getting the following errors and enum lookup doesn't work. Jun 19 11:29:41 WARNING[1218565440]: enum.c:186 parse_naptr: Regex compilation error (regex = !^+1613999$). Jun 19 11:29:41 WARNING[1218565440]: enum.c:264 enum_callback: Failed to parse naptr :( Jun 19 11:29:41 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to parse result Jun 19 11:29:41 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error I am just wondering if anyone expereinced similar problem, any suggestion will be appreciated. Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IRC
It seems the #asterisk channel on IRC has become an exclusive club. Suddenly it gives: An access level of [5] is required for [INVITE] on #asterisk irc://freenode/%23asterisk What's up? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problems with the latest CVS
I just recompiled * with the latest CVS. I am using enum in my extensions to dial first over the internet, if applicable. Everything was working perfectly, but now after installing the latest CVS I am getting the following errors and enum lookup doesn't work. Jun 19 11:29:41 WARNING[1218565440]: enum.c:186 parse_naptr: Regex compilation error (regex = !^+1613999$). Jun 19 11:29:41 WARNING[1218565440]: enum.c:264 enum_callback: Failed to parse naptr :( Jun 19 11:29:41 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to parse result Jun 19 11:29:41 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error I am just wondering if anyone expereinced similar problem, any suggestion will be appreciated. I could be totally wrong, but if memory serves correctly, I faintly remember seeing an asterisk-cvs entry that changed the default config options. Might look around in the cvs configs directory to see if that might be the case. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Thousands of contexts?
AGI script with SQL backend. You know the user (callerid) and the dialed number (exten) going into the script, and you return a variable that says yay or nay. -Original Message- From: Florian Overkamp [mailto:[EMAIL PROTECTED] Sent: Saturday, June 19, 2004 5:15 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Thousands of contexts? Hi, -Original Message- But the thought is correct, use a database to store the data and one context that does a lookup into the database and populates your callerid. It is a better way of doing things. You could even host it in the ast_db and then it shouldn't be too slow as you aren't spawning any outside apps. Here's another scenario I'm working with: I am using contexts per user to 'include' the numberranges they are allowed to dial. Any suggestions how to do that without a context per user ? Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
Its far from exclusive. I sent an email telling everyone what they must do. Its now +r which means you need to register with NickServ and Identify before you join. This was needed due to the spambots and the few abusive people. bkw - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 10:40 AM Subject: [Asterisk-Users] IRC It seems the #asterisk channel on IRC has become an exclusive club. Suddenly it gives: An access level of [5] is required for [INVITE] on #asterisk irc://freenode/%23asterisk What's up? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fw: [Asterisk-Users] #asterisk is +r now, meaning register your nick with nickserv
Here is the info again. bkw - Original Message - From: Brian K. West To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Friday, June 18, 2004 7:23 PM Subject: [Asterisk-Users] #asterisk is +r now, meaning register your nick with nickserv How do you register? do this /msg NickServ help or /msg NickServ register [yourpassword] You will be required to /msg NickServ IDENTIFY [yourpassword] before you can join #asterisk. I'm sorry we had to do this but the spambots that join and part 100+ times per hour were getting way out of hand. Thanks, Brian
Re: [Asterisk-Users] IRC
You need to be registered with nickserv to join. We had spam bots joining and leaving all day, and this fixes the problem. /msg nickserv register pass doughecka Steve Underwood wrote: It seems the #asterisk channel on IRC has become an exclusive club. Suddenly it gives: An access level of [5] is required for [INVITE] on #asterisk irc://freenode/%23asterisk What's up? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
??? Topic (#asterisk): The Asterisk Open Source PBX || Please register with nickserv to join #asterisk ??? Topic (#asterisk): set by kram at Fri Jun 18 14:27:17 2004 bkw - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 10:40 AM Subject: [Asterisk-Users] IRC It seems the #asterisk channel on IRC has become an exclusive club. Suddenly it gives: An access level of [5] is required for [INVITE] on #asterisk irc://freenode/%23asterisk What's up? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
Coppice is right. It is not enough to be registered with nickserv ... On Sat, 2004-06-19 at 18:27, Brian K. West wrote: Its far from exclusive. I sent an email telling everyone what they must do. Its now +r which means you need to register with NickServ and Identify before you join. This was needed due to the spambots and the few abusive people. bkw - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 10:40 AM Subject: [Asterisk-Users] IRC It seems the #asterisk channel on IRC has become an exclusive club. Suddenly it gives: An access level of [5] is required for [INVITE] on #asterisk irc://freenode/%23asterisk What's up? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hard Coded CLASS Codes (was 11 instead of Star)
In May, I posted an inquiry to the list concerning my desire to configure my own CLASS codes in extensions.conf rather than having them hard coded into the channel drivers. I have a number of old rotary dial telephones that (obviously) can't dial *. Traditionally in the US, 11 can be dialed in place of * as the first digit dialed. Many people mentioned that this would be very useful to them, especially those whose Asterisk PBXes are not located in the US, where their countrys' CLASS codes are different than the US standard. I guess I have to ask: If I want this change, as per the bug at http://bugs.digium.com/bug_view_page.php?bug_id=071 , What is the procedure? Do I just ask nicely and hope for the best? Or would funding be required to pay for having the code rewritten? If payment is necessary, how many of you members on the list would be willing to pitch in your own money to fund the effort? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Thousands of contexts?
Michael Bielicki wrote: Hi Sim :) you seem to be one of the top candidates as a ast_data power tester :) ast_data has gotten bloated beyond belief. All one needs is a proper understanding of the power of Asterisk's dial plan and you won't EVER have a need for thousands of contexts. Use what Mark has already given us, don't bastardize it with fluff. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: WaitExten substitute
At 6:47 AM -0700 on 6/19/04, Randy Bush wrote: I never heard of the app WaitExten http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20WaitExten but you could do the following: exten = _.,1,DigitTimeout,5 exten = _.,2,ResponseTimeout,10 exten = _.,3,SetVar(areacode=666) exten = _.,4,Background(zz-in-who) include = extensions include = applications include = speeddials exten = i,1,PlayBack(Invalid-Ext) exten = i,2,HangUp exten = t,1,PlayBack(NoExtensionEntered) exten = t,2,Hangup yes, i could; and i did. problem is that it does not seem to work; hence my posting. it plays the background and then falls through to invalid on the first keypress. i suspect this may be a sipura config issue again causing a double invite; but i am not sure. how do i hack this? Don't, I suggest you read the handbook, the wiki, and/or the archives. been there. done that. but thanks for the pointers. [ btw, search function in wiki is not real great, to be polite; but that issue is not local to this wiki ] randy I would suggest never using _. anywhere, since that matches i, t, h, and a host of other special extensions that are used internally in Asterisk's dialplan, which will result in unexpected extension handling. In many cases, the match _X. is what you want, since that matches at least one _digit_ and then any number of other digits. If that isn't quite what you want, then a combination of _X and _X. might work in some cascading fashion. This may not solve the problem you're having, but is a good syntax/logic habit to develop when creating complex dialplans. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Directory function is not working
Hello I am trying to setup Directory service for incoming call but it is not working. As per the document on voip-info.org, * should exist from directory but that is not happening. Is it a bug or I am doing something wrong? Please help. Thanks Deepak
Re: [Asterisk-Users] IRC
On Sat, 19 Jun 2004, Brian K. West wrote: Its far from exclusive. I sent an email telling everyone what they must do. Its now +r which means you need to register with NickServ and Identify before you join. This was needed due to the spambots and the few abusive people. Sounds like a bastardisation of what IRC is meant to be. IRCNET, you know it makes sense - no nickservs, no dodgy bots etc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
Hi, I figured it out. Most IRC channels requiring some authentication give a minute's latitude to allow for slow response from nickserv. It seems #asterisk is not doing that. You really must wait for nickserv to say you are registered before you issue a /join #asterisk. Regards, Steve Michael Bielicki wrote: Coppice is right. It is not enough to be registered with nickserv ... On Sat, 2004-06-19 at 18:27, Brian K. West wrote: Its far from exclusive. I sent an email telling everyone what they must do. Its now +r which means you need to register with NickServ and Identify before you join. This was needed due to the spambots and the few abusive people. bkw - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 10:40 AM Subject: [Asterisk-Users] IRC It seems the #asterisk channel on IRC has become an exclusive club. Suddenly it gives: An access level of [5] is required for [INVITE] on #asterisk irc://freenode/%23asterisk What's up? Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival and asterisk
i've try to remove the quote marks but still not working the * console
still return :
Executing Answer(SIP/4001-bf1a,) in new stack
Executing Festival(SIP/4001-bf1a,good morning) in new stack
Parsing '/etc/asterisk/festival.conf' : Found
Spawn extension (local,555,2) exited non-zero on 'SIP/4001-bf1a'
and the festival console return:
server Sun Jun 20 01:07:23 2004 : Festival server started on port 1314
client(1) Sun Jun 20 01:09:02 2004 : Accepted from localhost
client(1) Sun Jun 20 01:09:04 2004 : disconnected
its look like the festival does not process anything.
Best Regards,
Chiang
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson
Sent: Saturday, June 19, 2004 10:41 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Festival and asterisk
extension.conf
exten = 555,1,Answer
exten = 555,2,Festival('good morning')
exten = 555,3,Wait(2)
exten = 555,4,Hangup
What's the problem I'm facing? Thanks in advance.
Remove the quote marks... should be like ...Festival(good morning)
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Re: [Asterisk-Users] IRC
Steve Underwood wrote: Hi, I figured it out. Most IRC channels requiring some authentication give a minute's latitude to allow for slow response from nickserv. It seems #asterisk is not doing that. You really must wait for nickserv to say you are registered before you issue a /join #asterisk. You can't login to an ssh session until you are authenticated, why would IRC be any different? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
2 different things, you should be able to join a channel even if nickserv didn't authenticate you yet!!! but this is OT ;-))) Marc At 20:12 19.06.2004, you wrote: Steve Underwood wrote: Hi, I figured it out. Most IRC channels requiring some authentication give a minute's latitude to allow for slow response from nickserv. It seems #asterisk is not doing that. You really must wait for nickserv to say you are registered before you issue a /join #asterisk. You can't login to an ssh session until you are authenticated, why would IRC be any different? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
Damn dude, you are a worse troll than me... :P The fact is, freenode is so laggy with, but not restricted to, nickserv registrations... that the thing called perform in most irc clients is messed up, when using it to join some channels. Which can be useful when you are not on a very stable or 24/7 connection, such as dial-up or hong-kong based broadband connections ;) Other than that, it sucks to have it this way... but I know it is a problem with the spambots. Having the channel +s might be a better way to fight spambots, without putting a restriction on the users. Jeremy McNamara wrote: Steve Underwood wrote: Hi, I figured it out. Most IRC channels requiring some authentication give a minute's latitude to allow for slow response from nickserv. It seems #asterisk is not doing that. You really must wait for nickserv to say you are registered before you issue a /join #asterisk. You can't login to an ssh session until you are authenticated, why would IRC be any different? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
Michael Sandee wrote: Damn dude, you are a worse troll than me... :P The fact is, freenode is so laggy with, but not restricted to, nickserv registrations... that the thing called perform in most irc clients is messed up, when using it to join some channels. Which can be useful when you are not on a very stable or 24/7 connection, such as dial-up or hong-kong based broadband connections ;) Then people need to complain to freenode. Other than that, it sucks to have it this way... but I know it is a problem with the spambots. Having the channel +s might be a better way to fight spambots, without putting a restriction on the users. fight? they already know #asterisk exists. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
Its way toolate for +s bkw - Original Message - From: Michael Sandee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 12:41 PM Subject: Re: [Asterisk-Users] IRC Damn dude, you are a worse troll than me... :P The fact is, freenode is so laggy with, but not restricted to, nickserv registrations... that the thing called perform in most irc clients is messed up, when using it to join some channels. Which can be useful when you are not on a very stable or 24/7 connection, such as dial-up or hong-kong based broadband connections ;) Other than that, it sucks to have it this way... but I know it is a problem with the spambots. Having the channel +s might be a better way to fight spambots, without putting a restriction on the users. Jeremy McNamara wrote: Steve Underwood wrote: Hi, I figured it out. Most IRC channels requiring some authentication give a minute's latitude to allow for slow response from nickserv. It seems #asterisk is not doing that. You really must wait for nickserv to say you are registered before you issue a /join #asterisk. You can't login to an ssh session until you are authenticated, why would IRC be any different? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mediatrix 1204 Incoming calls
I would like to know if someone could help me when i recieved an incoming call on a Meditarix 1204 how to redirect the call? And the configuration i need? _Thanks -- ___ Get your free email from http://www.hackermail.com Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
Sorry to break your bubble... but: No it isn't. These bots come from owned hosts, and get the channel name from the server info which happily lists all non-secret channels. The operators actively battle the bots by k-lining them. (Or well, they are supposed to do this slave job for us atleast, it's the only thing they have to do apart from spying on us). These bots do NOT cache the channel names. They get them after connecting to the server. (Yes I know... we reversed some of them recently) P.S. Since you are a guy with ops YOU made the mistake by not having the channel +s in the first place. And thanks for making it +s now, lets hope it gets resolved now for the common good. Regards, Mike Brian K. West wrote: Its way toolate for +s bkw - Original Message - From: Michael Sandee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 12:41 PM Subject: Re: [Asterisk-Users] IRC Damn dude, you are a worse troll than me... :P The fact is, freenode is so laggy with, but not restricted to, nickserv registrations... that the thing called perform in most irc clients is messed up, when using it to join some channels. Which can be useful when you are not on a very stable or 24/7 connection, such as dial-up or hong-kong based broadband connections ;) Other than that, it sucks to have it this way... but I know it is a problem with the spambots. Having the channel +s might be a better way to fight spambots, without putting a restriction on the users. Jeremy McNamara wrote: Steve Underwood wrote: Hi, I figured it out. Most IRC channels requiring some authentication give a minute's latitude to allow for slow response from nickserv. It seems #asterisk is not doing that. You really must wait for nickserv to say you are registered before you issue a /join #asterisk. You can't login to an ssh session until you are authenticated, why would IRC be any different? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
Other than that, it sucks to have it this way... but I know it is a problem with the spambots. Having the channel +s might be a better way to fight spambots, without putting a restriction on the users. fight? they already know #asterisk exists. Aye, such big words for a guy who has to ask in the channel what +s means... I'll refer to my other reply for the details. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
We set +s for more than a day the spam bots kept coming... So +s wasn't working. We did -s so you could atleast see the channel topic telling you to register. I added it back today but that doesn't solve the problem. And you didn't break my bubble you just were not there to see that +s didn't do a damn thing. bkw - Original Message - From: Michael Sandee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 1:16 PM Subject: Re: [Asterisk-Users] IRC Sorry to break your bubble... but: No it isn't. These bots come from owned hosts, and get the channel name from the server info which happily lists all non-secret channels. The operators actively battle the bots by k-lining them. (Or well, they are supposed to do this slave job for us atleast, it's the only thing they have to do apart from spying on us). These bots do NOT cache the channel names. They get them after connecting to the server. (Yes I know... we reversed some of them recently) P.S. Since you are a guy with ops YOU made the mistake by not having the channel +s in the first place. And thanks for making it +s now, lets hope it gets resolved now for the common good. Regards, Mike Brian K. West wrote: Its way toolate for +s bkw - Original Message - From: Michael Sandee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 12:41 PM Subject: Re: [Asterisk-Users] IRC Damn dude, you are a worse troll than me... :P The fact is, freenode is so laggy with, but not restricted to, nickserv registrations... that the thing called perform in most irc clients is messed up, when using it to join some channels. Which can be useful when you are not on a very stable or 24/7 connection, such as dial-up or hong-kong based broadband connections ;) Other than that, it sucks to have it this way... but I know it is a problem with the spambots. Having the channel +s might be a better way to fight spambots, without putting a restriction on the users. Jeremy McNamara wrote: Steve Underwood wrote: Hi, I figured it out. Most IRC channels requiring some authentication give a minute's latitude to allow for slow response from nickserv. It seems #asterisk is not doing that. You really must wait for nickserv to say you are registered before you issue a /join #asterisk. You can't login to an ssh session until you are authenticated, why would IRC be any different? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival and asterisk
I think it should be without the quotes like this:
exten = 555,2,Festival(good morning)
- Original Message -
From: Freddy Setiawan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 19, 2004 1:50 AM
Subject: [Asterisk-Users] Festival and asterisk
I've install the asterisk in my Redhat 9 and it work properly with the SIP
phone. Then i install the festival as mention in
http://www.voip-info.org/wiki-Asterisk+festival+installation. the problem
is
when i dial the extension 555 in the asterisk console it show like :
-- Executing Answer(SIP/4001-664c, ) in new stack
Executing Festival(SIP/4001-664c, 'good morning') in new stack
Parsing '/etc/asterisk/festival.conf': Found
Spawn extension (local,555,2) exited non-zero on 'SIP/4001-664c'
and in the festival console it show :
server Sat Jun 19 13:37:28 2004 : Festival server started on port 1314
client(1) Sat Jun 19 13:50:03 2004 : accepted from localhost
client(1) Sat Jun 19 13:50:04 2004 : disconnected
*for references*
extension.conf
exten = 555,1,Answer
exten = 555,2,Festival('good morning')
exten = 555,3,Wait(2)
exten = 555,4,Hangup
What's the problem I'm facing? Thanks in advance.
Best regards,
Freddy Setiawan
::Simple is Everything, Nothing is Complex::
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RE: [Asterisk-Users] asterisk -rx not working well
On Sat, 2004-06-19 at 07:05, Sam Bingner wrote: It's exiting before the output finishes printing, it's a known bug and a timing issue There's a patch I put that's a hack to put in a timeout for exit, you should be able to find it on bugs.digium.com Instead of timing the exit, why don't you just flush the STDOUT and STDERR file descriptors? Seems this could be done before exit no matter what the reason was. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephan Wik Sent: Saturday, June 19, 2004 1:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk -rx not working well We're trying to use asterisk -r -x sip show peers to monitor sip phone availability. Sometimes the command shows the correct output but 9 times out of 10 all that is returned is: Name/usernameHost Mask Port Status with no listing. Anyone else see this behaviour? Stephan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRC
Well, it does do a damn thing in the long term. So keep it there. People who were not clueful enough to figure out the +r by themselves sure wouldn't have figured out the topic either... ;) (Sorry for all the troll/flame, but the tread started with a silly reply with a stupid analogy.) Brian K. West wrote: We set +s for more than a day the spam bots kept coming... So +s wasn't working. We did -s so you could atleast see the channel topic telling you to register. I added it back today but that doesn't solve the problem. And you didn't break my bubble you just were not there to see that +s didn't do a damn thing. bkw - Original Message - From: Michael Sandee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 1:16 PM Subject: Re: [Asterisk-Users] IRC Sorry to break your bubble... but: No it isn't. These bots come from owned hosts, and get the channel name from the server info which happily lists all non-secret channels. The operators actively battle the bots by k-lining them. (Or well, they are supposed to do this slave job for us atleast, it's the only thing they have to do apart from spying on us). These bots do NOT cache the channel names. They get them after connecting to the server. (Yes I know... we reversed some of them recently) P.S. Since you are a guy with ops YOU made the mistake by not having the channel +s in the first place. And thanks for making it +s now, lets hope it gets resolved now for the common good. Regards, Mike Brian K. West wrote: Its way toolate for +s bkw - Original Message - From: Michael Sandee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 12:41 PM Subject: Re: [Asterisk-Users] IRC Damn dude, you are a worse troll than me... :P The fact is, freenode is so laggy with, but not restricted to, nickserv registrations... that the thing called perform in most irc clients is messed up, when using it to join some channels. Which can be useful when you are not on a very stable or 24/7 connection, such as dial-up or hong-kong based broadband connections ;) Other than that, it sucks to have it this way... but I know it is a problem with the spambots. Having the channel +s might be a better way to fight spambots, without putting a restriction on the users. Jeremy McNamara wrote: Steve Underwood wrote: Hi, I figured it out. Most IRC channels requiring some authentication give a minute's latitude to allow for slow response from nickserv. It seems #asterisk is not doing that. You really must wait for nickserv to say you are registered before you issue a /join #asterisk. You can't login to an ssh session until you are authenticated, why would IRC be any different? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] enum problems with the latest CVS
Please ignore this message, everything is back to normal :) Wojtek - Original Message - From: Wojciech Tryc [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 11:35 AM Subject: [Asterisk-Users] enum problems with the latest CVS Hi, I just recompiled * with the latest CVS. I am using enum in my extensions to dial first over the internet, if applicable. Everything was working perfectly, but now after installing the latest CVS I am getting the following errors and enum lookup doesn't work. Jun 19 11:29:41 WARNING[1218565440]: enum.c:186 parse_naptr: Regex compilation error (regex = !^+1613999$). Jun 19 11:29:41 WARNING[1218565440]: enum.c:264 enum_callback: Failed to parse naptr :( Jun 19 11:29:41 WARNING[1218565440]: dns.c:141 dns_parse_answer: Failed to parse result Jun 19 11:29:41 WARNING[1218565440]: dns.c:183 ast_search_dns: Parse error I am just wondering if anyone expereinced similar problem, any suggestion will be appreciated. Regards, Wojtek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HST Saphir with Asterisk
Hello, I would like to use my existing HST Saphir V S2M PCI with Asterisk. Unfortunately I could not find any information about the usage with Asterisk. Does someone know if it's possible to use it? Do I need to do something special? Regards Julian Pawlowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HST Saphir with Asterisk
On Saturday 19 June 2004 17:59, Julian Pawlowski wrote: I would like to use my existing HST Saphir V S2M PCI with Asterisk. Unfortunately I could not find any information about the usage with Asterisk. You're on your own; as a general rule the Asterisk community has turned up their nose at trying to get every brand of modem imaginable working properly with Asterisk. There are just too many variables -- is the chipset documentation available? Is the card capable of full-duplex operation? Does it have a high-quality hybrid? The list just goes on, and I for one agree with Digium on this -- You're more than welcome to try to get it to work but it's not supported. The official recommendation is to purchase an X101P from Digium. If you're resourceful (hint: use google and search the mailing list archives) you will find that it is possible to obtain the same hardware for much more economically, although it is officially unsupported and others have had trouble buying X101P clones. Digium also sells a kit which has one port to plug a phone into and one port to plug a phone line in to as an entry level kit. This kit is expandable, although it's not exactly cheap. You must remember that Asterisk is provided completely free of charge for noncommercial use; it's not too much to ask to buy the hardware from Digium. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk -rx not working well
The problem is that it uses a socket, and doesn't KNOW when the end of the command is. Somebody else IS working on fixing this so there is a control connection and it will know. It does finish writing everything that the remote console got from the * server before it exits... It just hasn't got everything yet. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Saturday, June 19, 2004 10:17 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk -rx not working well On Sat, 2004-06-19 at 07:05, Sam Bingner wrote: It's exiting before the output finishes printing, it's a known bug and a timing issue There's a patch I put that's a hack to put in a timeout for exit, you should be able to find it on bugs.digium.com Instead of timing the exit, why don't you just flush the STDOUT and STDERR file descriptors? Seems this could be done before exit no matter what the reason was. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephan Wik Sent: Saturday, June 19, 2004 1:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk -rx not working well We're trying to use asterisk -r -x sip show peers to monitor sip phone availability. Sometimes the command shows the correct output but 9 times out of 10 all that is returned is: Name/usernameHost Mask Port Status with no listing. Anyone else see this behaviour? Stephan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
Re: [Asterisk-Users] HST Saphir with Asterisk
Hi Il dom, 2004-06-20 alle 00:21, Andrew Kohlsmith ha scritto: On Saturday 19 June 2004 17:59, Julian Pawlowski wrote: I would like to use my existing HST Saphir V S2M PCI with Asterisk. Unfortunately I could not find any information about the usage with Asterisk. snip The official recommendation is to purchase an X101P from Digium. If you're resourceful (hint: use google and search the mailing list archives) you will find that it is possible to obtain the same hardware for much more economically, although it is officially unsupported and others have had trouble buying X101P clones. the card in discussion is a isdn E1 pri card. HST is listed as capi hardware manufacturer under www.capi.org you should be able to use that with kapejod wonderful chan_capi and kernel drivers from http://www.hstnet.de/english/downloads/isdn/saphir_5_primary_pci/index.asp unfortunately they provide only precompiled kernel modules, and only for certain kernel versions... so you're stick to what they use. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HST Saphir with Asterisk
On Saturday 19 June 2004 18:59, Brancaleoni Matteo wrote: the card in discussion is a isdn E1 pri card. ugh. I apologize. I saw HST and figured it was a variant of the HCF/HSF nonsense of Winmodems. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Thousands of contexts?
Jeremy McNamara wrote: Michael Bielicki wrote: Hi Sim :) you seem to be one of the top candidates as a ast_data power tester :) ast_data has gotten bloated beyond belief. All one needs is a proper understanding of the power of Asterisk's dial plan and you won't EVER have a need for thousands of contexts. Use what Mark has already given us, don't bastardize it with fluff. Very well said!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFax problems
Hey All, I'm still (since April) having problems getting RxFax to work over an ISDN4Linux channel. Just wondering if anyone has had any luck getting it to work? I have done a CVS update today (about half hour ago) and made sure I have the latest version of spandsp according to Steve's website (spandsp-0.0.1k). When I was compiling asterisk, I got the following warnings: == gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/08/04-10:06:15\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c -o app_rxfax.o app_rxfax.c In file included from ../include/spandsp.h:40, from app_rxfax.c:29: ../include/spandsp/arctan2.h: In function `arctan2': ../include/spandsp/arctan2.h:51: warning: implicit declaration of function `fabs' In file included from ../include/spandsp.h:47, from app_rxfax.c:29: ../include/spandsp/dc_restore.h: In function `fsaturate': ../include/spandsp/dc_restore.h:105: warning: implicit declaration of function `lrint' app_rxfax.c: At top level: app_rxfax.c:50: warning: no previous prototype for `t30_flush' app_rxfax.c:57: warning: no previous prototype for `phase_e_handler' gcc -shared -Xlinker -x -I/usr/src/spandsp-0.0.1/src -o app_rxfax.so app_rxfax.o -L/usr/src/spandsp-0.0.1/src -lspandsp -ltiff gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/08/04-10:06:15\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c -o app_txfax.o app_txfax.c In file included from ../include/spandsp.h:40, from app_txfax.c:27: ../include/spandsp/arctan2.h: In function `arctan2': ../include/spandsp/arctan2.h:51: warning: implicit declaration of function `fabs' In file included from ../include/spandsp.h:47, from app_txfax.c:27: ../include/spandsp/dc_restore.h: In function `fsaturate': ../include/spandsp/dc_restore.h:105: warning: implicit declaration of function `lrint' app_txfax.c: At top level: app_txfax.c:46: warning: no previous prototype for `t30_flush' app_txfax.c:52: warning: no previous prototype for `phase_e_handler' gcc -shared -Xlinker -x -I/usr/src/spandsp-0.0.1/src -o app_txfax.so app_txfax.o -L/usr/src/spandsp-0.0.1/src -lspandsp -ltiff == Although it continued to compile without any other errors or warnings. When I try to receive a fax (sent from a Rockwell HCF modem) I get the following output in the asterisk console, and the originating fax doesn't handshake with it: == -- Executing Goto(Modem[i4l]/ttyI0, fax|s|1) in new stack -- Goto (fax,s,1) -- Executing Macro(Modem[i4l]/ttyI0, faxreceive) in new stack -- Executing RxFAX(Modem[i4l]/ttyI0, /var/spool/asterisk-fax/1087693402.1.tif) in new stack Jun 20 10:34:25 NOTICE[294927]: channel.c:1651 ast_set_read_format: Unable to find a path from SLINR to UNKN Jun 20 10:34:25 WARNING[294927]: app_rxfax.c:253 rxfax_exec: Unable to restore read format on 'Modem[i4l]/ttyI0' Jun 20 10:34:25 NOTICE[294927]: channel.c:1618 ast_set_write_format: Unable to find a path from UNKN to SLINR Jun 20 10:34:25 WARNING[294927]: app_rxfax.c:259 rxfax_exec: Unable to restore write format on 'Modem[i4l]/ttyI0' == Spawn extension (macro-faxreceive, s, 1) exited non-zero on 'Modem[i4l]/ttyI0' in macro 'faxreceive' == Spawn extension (fax, s, 1) exited non-zero on 'Modem[i4l]/ttyI0' -- Hungup 'Modem[i4l]/ttyI0' == Does anyone have any idea how I can get this to work? At the moment I am only really interested in receiving faxes, but sending might be nice in the future. Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IRC
You need to be registered with nickserv to join. We had spam bots joining and leaving all day, and this fixes the problem. No - like painkillers this might remove the symptoms - it most definitely does not fix the problem. //Lars ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival and asterisk
On Sat, Jun 19, 2004 at 08:40:36AM -0600, Rich Adamson wrote:
extension.conf
exten = 555,1,Answer
exten = 555,2,Festival('good morning')
exten = 555,3,Wait(2)
exten = 555,4,Hangup
What's the problem I'm facing? Thanks in advance.
Remove the quote marks... should be like ...Festival(good morning)
What is the difference between an argument with quotes, and one without?
I ask because on one page of the wiki, it says not to use them, yet on
another [2] it highlights the use of quotes.
I have installed Festival today and it has been working without quotes,
but I wonder about whether it will parse (and change inflection) if
there are commas, question marks, and or exclamation points present and,
if so, if having them in the argument to Festival() will cause issues
without quotes.
[1] http://www.voip-info.org/wiki-Asterisk_Festival_installation
[2] http://www.voip-info.org/wiki-Asterisk+cmd+Festival
SWS
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[Asterisk-Users] Maximum retries exceeded w/SIP
I struggled with this for several hours tonight.Turns out that if you have an * machine behind NAT, you must put the PUBLIC address in the bindaddr in sip.confIf you don't put it in, the Contact: header contains the NATted address and the sip phone can't get back to *.I don't know what happens if you mix and match sip phones on the local network -- it might not work unless the sipphone uses the public address as well.Hope this helps as I see this thread come up again and again...Andy---Steve, Sure, I could put all my machines on the public Internet, but that defeats the purpose of having a firewall in the first place. As an alternative, I could only place the * server on the outside, but I'd rather not give the script-kiddies another box to pound. Steve Totaro wrote: Can you disable your firewall? i am about to start this phase of asterisk an would like help from one newbie to another. otherwise this newbie will let you know how i did it. - Original Message - From: "Brad Waite" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 20, 2003 9:07 AM Subject: [Asterisk-Users] Maximum retries exceeded w/SIP First of all, I'd like to send a big "thank you" to all the folks who have helped me get this far. Now on to the next problem. Here's my current network setup: The Big I ---+--- FreeBSD FW --- * (10.0.0.253) PC (10.0.0.1) | +--- Laptop (public IP) natd is set up with the following rules: redirect_port udp 10.0.0.253:1-2 1-2 redirect_port udp 10.0.0.253:5060 5060 * is set up with the demo/sandbox config. I'm using XLite as my SIP client and have configured it on PC to work with *. I'm able to do everything I've tried so far. I should, though - I'm on the inside. However, when trying to make a call from the outside (via Laptop), something's breaking. I've set up the SIP proxy in XLite to be the external interface on the firewall, and am able to log into the proxy without difficulty. And while I can begin conversations, I can't keep them going for long. For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I get most of the "demo-abouttotry" message - "I am about to attempt an IAX connection to a demonstration server located at Di" - at which point it gets cut off. The console spits out the following error: File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 12384 (Response) Any ideas what could be going on? My first guess is the firewall, but I can't figure out why some of the packets would get through while others apparently are not. I'm at a loss. Brad Waite aka HankPoacher ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Maximum retries exceeded w/SIP
Usage of externip= and localnet= are what you are looking for. These all have been covered more than once in the mailing list... Remember GOOGLE IS YOUR FRIEND!! :P bkw - Original Message - From: Andrew Sackheim To: [EMAIL PROTECTED] Sent: Saturday, June 19, 2004 9:29 PM Subject: [Asterisk-Users] Maximum retries exceeded w/SIP I struggled with this for several hours tonight.Turns out that if you have an * machine behind NAT, you must put the PUBLIC address in the bindaddr in sip.confIf you don't put it in, the Contact: header contains the NATted address and the sip phone can't get back to *.I don't know what happens if you mix and match sip phones on the local network -- it might not work unless the sipphone uses the public address as well.Hope this helps as I see this thread come up again and again...Andy---Steve, Sure, I could put all my machines on the public Internet, but that defeats the purpose of having a firewall in the first place. As an alternative, I could only place the * server on the outside, but I'd rather not give the script-kiddies another box to pound. Steve Totaro wrote: Can you disable your firewall? i am about to start this phase of asterisk an would like help from one newbie to another. otherwise this newbie will let you know how i did it. - Original Message - From: "Brad Waite" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 20, 2003 9:07 AM Subject: [Asterisk-Users] Maximum retries exceeded w/SIP First of all, I'd like to send a big "thank you" to all the folks who have helped me get this far. Now on to the next problem. Here's my current network setup: The Big I ---+--- FreeBSD FW --- * (10.0.0.253) PC (10.0.0.1) | +--- Laptop (public IP) natd is set up with the following rules: redirect_port udp 10.0.0.253:1-2 1-2 redirect_port udp 10.0.0.253:5060 5060 * is set up with the demo/sandbox config. I'm using XLite as my SIP client and have configured it on PC to work with *. I'm able to do everything I've tried so far. I should, though - I'm on the inside. However, when trying to make a call from the outside (via Laptop), something's breaking. I've set up the SIP proxy in XLite to be the external interface on the firewall, and am able to log into the proxy without difficulty. And while I can begin conversations, I can't keep them going for long. For instance, when trying to call [EMAIL PROTECTED] (or [EMAIL PROTECTED]), I get most of the "demo-abouttotry" message - "I am about to attempt an IAX connection to a demonstration server located at Di" - at which point it gets cut off. The console spits out the following error: File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 12384 (Response) Any ideas what could be going on? My first guess is the firewall, but I can't figure out why some of the packets would get through while others apparently are not. I'm at a loss. Brad Waite aka HankPoacher ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
