Re: [Asterisk-Users] billing realtime
random cluster wrote: Now, the question, can I access somehow in a deadagi, or whatever the CDR function in order to update the credit when the call has just finished. Yes, certainly, through deadagi. I just have one question though, why reinvent the wheel? There are prepaid systems that work with asterisk. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] Background asynchronous AGI
Matt wrote: Can't you do all of this with the (Absolute) time setting? So if the person has 4,000 minutes left.. set the call length for 4,000 minutes as the absolute max. Alternately... you could probably use screen? Launch an AGI from the main AGI using screen so it goes into the background... Or like astcc does, use the L argument to the dial command. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] billing realtime
JP Carballo wrote: Yes, certainly, through deadagi. I just have one question though, why reinvent the wheel? There are prepaid systems that work with asterisk. I have yet to find a prepaid system that allows multiple concurrent calls per account. Most seem to be based on a pin number also which I don't want. Anyone know of a system that allows concurrent calls? -- Jon Farmer Telford, Shropshire, UK ___ --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] TDM400P: flash on analog phones doesn't work
Patrick wrote: Hi, I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? snip Patrick, Can you check your flash button interval? Some phones produce a very short pulse when you press the button, which may not be detectable by the PBX. You can adjust the flash timing in zapata.conf ___ --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] billing realtime
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jon, we can do that using ASTPP. The downside is that we don't currently have a way to limit the call lengths so that when they have multiple calls in progress they still can't go over their prepaid limit. On postpaid accounts this is not usually an issue but on prepaid it still is. Darren Wiebe [EMAIL PROTECTED] Jon Farmer wrote: JP Carballo wrote: Yes, certainly, through deadagi. I just have one question though, why reinvent the wheel? There are prepaid systems that work with asterisk. I have yet to find a prepaid system that allows multiple concurrent calls per account. Most seem to be based on a pin number also which I don't want. Anyone know of a system that allows concurrent calls? -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFETxRg4DADnh+tnOQRAuhJAJ9kzGiQYh4Z6WPXXes6TKtwusBliwCeMvHG 3nrqsxdXNrfJbCZ3uzlpd5w= =+fV+ -END PGP SIGNATURE- ___ --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] Need some help on queues with agents(SIP members) with multiple phones.
Hi. We have people with two or more sip phones. One wireless and one wired. So this is the case: Person A with two phones wants to have a queue for his incoming calls. So when he answers one of the two phones, the other phone should not ring. But when he isn't talking in any of the phones, they both should ring. Does that make any sense? This what I have for people with only one Sip-phone: Customer calls in. Dials SIP Phone. If answered. OK. Else If BUSY go to queue. Else go to voicemail. How would I manage this for the situation described above? Dialplan examples would be appreciated. Regards, Arne Morten Johansen ___ --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] billing realtime
On Wed April 26 2006 16:31, Jon Farmer [EMAIL PROTECTED] wrote: JP Carballo wrote: Yes, certainly, through deadagi. I just have one question though, why reinvent the wheel? There are prepaid systems that work with asterisk. I have yet to find a prepaid system that allows multiple concurrent calls per account. Most seem to be based on a pin number also which I don't want. Anyone know of a system that allows concurrent calls? -- Jon Farmer Telford, Shropshire, UK Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 concurrent calls, how do you know to cut off the 2 calls at the 5 minute mark rather than cut off both calls after 10 minutes? -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --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] AGI and incoming call
Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.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
SV: [Asterisk-Users] Need some help on queues with agents(SIP members)with multiple phones.
I also have some other trouble. How the I send the caller to voicemail (next extension) if the Member = SIP/phone stops answering for a defined period of time. I cant figure out if this would work (from queues.conf): ; If you wish to remove callers from the queue if there are no agents present, then set ; this to yes. Note that this is for use with dynamic queue members! ; ; leavewhenempty = yes -Opprinnelig melding- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Arne Morten Johansen Sendt: 26. april 2006 08:41 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Need some help on queues with agents(SIP members)with multiple phones. Hi. We have people with two or more sip phones. One wireless and one wired. So this is the case: Person A with two phones wants to have a queue for his incoming calls. So when he answers one of the two phones, the other phone should not ring. But when he isn't talking in any of the phones, they both should ring. Does that make any sense? This what I have for people with only one Sip-phone: Customer calls in. Dials SIP Phone. If answered. OK. Else If BUSY go to queue. Else go to voicemail. How would I manage this for the situation described above? Dialplan examples would be appreciated. Regards, Arne Morten Johansen ___ --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] billing realtime
Nick Hoffman wrote: Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 concurrent calls, how do you know to cut off the 2 calls at the 5 minute mark rather than cut off both calls after 10 minutes? -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. There's an application (sorry, which one, escapes me at the moment), that gets around this by reserving a certain amount of credit per call. Say the amount is 10 minutes, if you have 30 minutes worth of credit, you can have 3 concurrent calls good for 10 minutes each. The way I understand it, if you only have 15 minutes left in your account, the first call will last for 10 and the next concurrent one for 5 minutes. -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ --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] billing realtime
[EMAIL PROTECTED] wrote: On Wed April 26 2006 16:31, Jon Farmer [EMAIL PROTECTED] wrote: JP Carballo wrote: Yes, certainly, through deadagi. I just have one question though, why reinvent the wheel? There are prepaid systems that work with asterisk. I have yet to find a prepaid system that allows multiple concurrent calls per account. Most seem to be based on a pin number also which I don't want. Anyone know of a system that allows concurrent calls? -- Jon Farmer Telford, Shropshire, UK Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 concurrent calls, how do you know to cut off the 2 calls at the 5 minute mark rather than cut off both calls after 10 minutes? The way we solved this is: 1/ Each account has incoming/outgoing channels 2/ Once call is started then the total balance is divided by number of outgoing channels for that account. This sets the time limit. 3/ If more calls are made then each new call has same absolute timeout. Above is not perfect, since we are limiting each call to less talk time then total balance allows, hence why we are currently looking into possibility in changing the value of absolute timeout in memory for each of the calls. Senad ___ --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] CISCO 7960G - SIP Configuration
Hello all, Just got hold of a CISCO 7960G. Updated the Firmware to the latest 8.2. As you all know it has 6 lines which is why i bought it. Just would like to know from you experts if this piece will connect to 6 different providers over the internet or will it only work as 6 extns with 1 provider. Im not able to get connected to more than one provider. Also i have found that the time is not working well. It runs at 1 hour ahead even if Time Zone is set correctly. Need some advice. Thanks Dan ___ --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] # and call speed
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, if I append a # after the number asterisk call more fastly, but which step I bypass? Can I append this in all call automaticaly? If yes, how can I do this? - -- Pasqualotto Enrico email: [EMAIL PROTECTED] web: http://www.pasqualotto.org - -BEGIN GEEK CODE BLOCK- Version: 3.12 GIT d? s: a-- C+++ UL P L++ E--- W++ N++ o K- w--- O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+ G e h++ r+ y+ - --END GEEK CODE BLOCK-- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFETyY7c25ND+sg2LkRAmSjAJ9TZPAk51OL2u7nwhQHfrtCRYt3sQCgi2KF 2wdSh8JLkyLgKgf53T1m+S0= =Qk2a -END PGP SIGNATURE- ___ --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] Intergrate Asterisk IP PBX with Legacy PBX, continuing existing funtionality of legacy pbx
Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story This is with regard to the setup which you can find at the Asterisk The Future of Telephony , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our Conventional PABX system. *Existing Setup over view* Our existing includes traditional Pabx, E1 Line from telecom provider, 16 direct lines another telecom provider. there are around 120 extensions. E1 Link using for DID and 16 lines using for as hunting group. *New Integration.* Integrate asterisk ip PBX with legacy Pabx which continues functionality of the existing setup I am planning to install 2 E1 cards in Asterisk box. Remove E1 link from legacy Pabx and fix it to 1 E1 card and other E1 card will using to connect traditional PABX. All previous DID's which configured with the traditional PABX will be configured in asterisk. Actually I am not sure that i will be able to achive this migration , but i am trying to acomplish. It is very much appreciate that if anyone can guide me on this regard. Thanks Regards, Vidura Senadeera. Sri Lanka. attachment: legacy_pbx_to_asterisk_migration.JPG ___ --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: billing realtime
In article [EMAIL PROTECTED], Darren Wiebe [EMAIL PROTECTED] wrote: The downside is that we don't currently have a way to limit the call lengths so that when they have multiple calls in progress they still can't go over their prepaid limit. This is exactly the problem I am trying to solve I could probably do it with a separate supervisor program using the Manager API, but can't think of any way to do it with AGI. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 and incoming call
Thats how I do it. AGI makes a call to a MSSQL server. A stored proc returns an extension based on ANI and DNIS and the call continues. Thanks, Steve Totaro Olivier Saulnier wrote: Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, ___ --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: billing realtime
In article [EMAIL PROTECTED], Senad Jordanovic [EMAIL PROTECTED] wrote: The way we solved this is: 1/ Each account has incoming/outgoing channels 2/ Once call is started then the total balance is divided by number of outgoing channels for that account. This sets the time limit. 3/ If more calls are made then each new call has same absolute timeout. Above is not perfect, since we are limiting each call to less talk time then total balance allows, hence why we are currently looking into possibility in changing the value of absolute timeout in memory for each of the calls. The other situation to take account of is when the caller somehow adds to his prepaid balance while he has one or more calls in progress, in order to avoid being cut off during the call. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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 and incoming call
Could you help me for AGI script and interface with asterisk server, please? Best regards, OLS Steve Totaro a écrit : Thats how I do it. AGI makes a call to a MSSQL server. A stored proc returns an extension based on ANI and DNIS and the call continues. Thanks, Steve Totaro Olivier Saulnier wrote: Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, ___ --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 -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.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] billing realtime
Hi Senad
i looking for same thing, that is consider absolutetimeout as a
timer, everytime is near t zero, 3 secs for example, renew it,
reacalculate real credit, and start again until some of the parties
hangup.
The problem is how to iterate in asterisk config, or in deadagi,
you will need some time values from asterisk anyway, CDR{billsec} and
CDR{duration}, because i think we have to give this control to
asterisk, he really knows the timing of calls. Now the problem number
two. Asterisk set those values above, when the call is completely
finished, i have tried with deadagi in php whit sleep function,
nothing, the values of the varialbles are set after hangup extension,
after deadagi final execution.
The solution that I looking for is to take a average-time-call, and
create a timer with it.
Then base on this value, and the price for destination call, every
time the average-time-call pass substract the consume credit from the
real credit, and set absolute timeout, for this average-time-call.
But I dont know how to implement this is asterisk. With pseudo-code
while
2006/4/26, Senad Jordanovic [EMAIL PROTECTED]:
[EMAIL PROTECTED] wrote:
On Wed April 26 2006 16:31, Jon Farmer [EMAIL PROTECTED] wrote:
JP Carballo wrote:
Yes, certainly, through deadagi.
I just have one question though, why reinvent the wheel?
There are prepaid systems that work with asterisk.
I have yet to find a prepaid system that allows multiple concurrent
calls per account. Most seem to be based on a pin number also which I
don't want. Anyone know of a system that allows concurrent calls? --
Jon Farmer
Telford, Shropshire, UK
Hi Jon. If a customer has 10 minutes of call credit left and he makes
2 concurrent calls, how do you know to cut off the 2 calls at the 5
minute mark rather than cut off both calls after 10 minutes?
The way we solved this is:
1/ Each account has incoming/outgoing channels
2/ Once call is started then the total balance is divided by number of
outgoing channels for that account. This sets the time limit.
3/ If more calls are made then each new call has same absolute timeout.
Above is not perfect, since we are limiting each call to less talk time then
total balance allows, hence why we are currently
looking into possibility in changing the value of absolute timeout in memory
for each of the calls.
Senad
___
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[Asterisk-Users] Stange behaviour on 4port BRI Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q
---BeginMessage--- Hi, we currently run Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q with 3 isdn-2 channels in euroisdn mode. the uptime on the box has been 94 days without problems but out of the blue it looks like the calls have been cutting out after 10-15 seconds, both inbound and outbound. nothing was logged in /var/log/messages about it or /var/log/asterisk/messages it looks like two zap channels were in use and the rest behaved in this manner. anyone else had this problem or can point me in the right direction of diagnosing what happened. -- Regards, Alasdair Gow BSc (Hons) Support Specialist Colloquium Internet Support ---End Message--- ___ --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] Avoiding deadlock... Problem
Hi I have 3FXO trunks called ZAP-25,ZAP-26 and ZAP-28 and T1 Channnel bank I get this deadlock problem when 2 incoming call from FXO(Here ZAP-28 and then ZAP-26) wants to dial same channel (Here ZAP-1). In this senario ZAP-1 first answer ZAP-28 and thne ZAP-26 wants to call ZAP-1 but it time out and goto voicemail after that ZAP-1 try to reach ZAP-26 call by puting ZAP-28 on HOLD During this period this this Notice is generates. And sometimes because of this Lines goes to dead. and need to restart asterisk. Please help me. Here is my LOG --- Apr 25 16:39:53 VERBOSE[3514] logger.c: -- Starting simple switch on 'Zap/28-1' Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Executing Set(Zap/28-1, FROM=s) in new stack Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Executing Goto(Zap/28-1, incoming-ivr|s|1) in new stack Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Goto (incoming-ivr,s,1) Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Executing GotoIf(Zap/28-1, 1?3) in new stack Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Goto (incoming-ivr,s,3) Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Executing Answer(Zap/28-1, ) in new stack Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Executing Set(Zap/28-1, TIMEOUT(digit)=5) in new stack Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Digit timeout set to 5 Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Executing Set(Zap/28-1, TIMEOUT(response)=7) in new stack Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Response timeout set to 7 Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Executing BackGround(Zap/28-1, silence/1) in new stack Apr 25 16:39:54 VERBOSE[3514] logger.c: -- Playing 'silence/1' (language 'en') Apr 25 16:39:55 VERBOSE[3514] logger.c: -- Executing BackGround(Zap/28-1, maingreeting) in new stack Apr 25 16:39:55 VERBOSE[3514] logger.c: -- Playing 'maingreeting' (language 'en') Apr 25 16:40:03 VERBOSE[3530] logger.c: -- Starting simple switch on 'Zap/26-1' Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Executing Set(Zap/26-1, FROM=s) in new stack Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Executing Goto(Zap/26-1, incoming-ivr|s|1) in new stack Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Goto (incoming-ivr,s,1) Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Executing GotoIf(Zap/26-1, 1?3) in new stack Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Goto (incoming-ivr,s,3) Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Executing Answer(Zap/26-1, ) in new stack Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Executing Set(Zap/26-1, TIMEOUT(digit)=5) in new stack Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Digit timeout set to 5 Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Executing Set(Zap/26-1, TIMEOUT(response)=7) in new stack Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Response timeout set to 7 Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Executing BackGround(Zap/26-1, silence/1) in new stack Apr 25 16:40:04 VERBOSE[3530] logger.c: -- Playing 'silence/1' (language 'en') Apr 25 16:40:05 VERBOSE[3530] logger.c: -- Executing BackGround(Zap/26-1, maingreeting) in new stack Apr 25 16:40:05 VERBOSE[3530] logger.c: -- Playing 'maingreeting' (language 'en') Apr 25 16:40:06 VERBOSE[3514] logger.c: == CDR updated on Zap/28-1 Apr 25 16:40:06 VERBOSE[3514] logger.c: -- Executing Macro(Zap/28-1, dial|ZAP/1|101) in new stack Apr 25 16:40:06 VERBOSE[3514] logger.c: -- Executing Dial(Zap/28-1, ZAP/1|15|) in new stack Apr 25 16:40:06 VERBOSE[3514] logger.c: -- Called 1 Apr 25 16:40:06 VERBOSE[3514] logger.c: -- Zap/1-1 is ringing Apr 25 16:40:08 VERBOSE[3514] logger.c: -- Zap/1-1 is ringing Apr 25 16:40:12 VERBOSE[3514] logger.c: -- Zap/1-1 answered Zap/28-1 Apr 25 16:40:12 VERBOSE[3514] logger.c: -- Attempting native bridge of Zap/28-1 and Zap/1-1 Apr 25 16:40:18 VERBOSE[3530] logger.c: == CDR updated on Zap/26-1 Apr 25 16:40:18 VERBOSE[3530] logger.c: -- Executing Macro(Zap/26-1, dial|ZAP/1|101) in new stack Apr 25 16:40:18 VERBOSE[3530] logger.c: -- Executing Dial(Zap/26-1, ZAP/1|15|) in new stack Apr 25 16:40:18 VERBOSE[3530] logger.c: -- Called 1 Apr 25 16:40:19 VERBOSE[3530] logger.c: -- Zap/1-2 is ringing Apr 25 16:40:19 VERBOSE[3514] logger.c: -- CPE does not support Call Waiting Caller*ID. Apr 25 16:40:34 VERBOSE[3530] logger.c: -- Nobody picked up in 15000 ms Apr 25 16:40:34 VERBOSE[3530] logger.c: -- Hungup 'Zap/1-2' Apr 25 16:40:34 VERBOSE[3530] logger.c: -- Executing GotoIf(Zap/26-1, 0?s-NOANSWER|1) in new stack Apr 25 16:40:34 VERBOSE[3530] logger.c: -- Executing Macro(Zap/26-1, vm|101|NOANSWER) in new stack Apr 25 16:40:34 VERBOSE[3530] logger.c: -- Executing Goto(Zap/26-1, s-NOANSWER|1) in new stack Apr 25 16:40:34 VERBOSE[3530] logger.c: -- Goto (macro-vm,s-NOANSWER,1) Apr 25 16:40:34 VERBOSE[3530] logger.c: -- Executing VoiceMail(Zap/26-1, u101) in new stack Apr 25 16:40:34 VERBOSE[3530] logger.c: -- Playing
RE: [Asterisk-Users] Re: billing realtime
[EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Senad Jordanovic [EMAIL PROTECTED] wrote: The way we solved this is: 1/ Each account has incoming/outgoing channels 2/ Once call is started then the total balance is divided by number of outgoing channels for that account. This sets the time limit. 3/ If more calls are made then each new call has same absolute timeout. Above is not perfect, since we are limiting each call to less talk time then total balance allows, hence why we are currently looking into possibility in changing the value of absolute timeout in memory for each of the calls. The other situation to take account of is when the caller somehow adds to his prepaid balance while he has one or more calls in progress, in order to avoid being cut off during the call. Noted!!! Thanks :) Senad ___ --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] test numbers in different countries!
On Wednesday 26 April 2006 07:52, Jason Frisch wrote: How about using time announments? I list of these for each country would be great! I have some test numbers on my switch in Latvia: +371 7160201 -- echo +371 7160202 -- music :) +371 7160203 -- time Do you mean something like 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
[Asterisk-Users] Excessive Asterisk delay to answer on ZAP inbound call
Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __Asterisk fxo line -| -Analog phone The analog phone rings immediately when calling, while asterisk shows the message Starting simple switch on zap... after the first ring and executes the old extension script after the second ring (for example a NoOp instruction). Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? TIA Giorgio Incantalupo ___ --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] test numbers in different countries!
In the UK you can use: 17070 on BT phone lines (UK) - you can then carry out a ring back test (option 1). Hope that's what you are looking for, and no-one has posted it before :) -Original Message- From: Dmitry Ivanov [mailto:[EMAIL PROTECTED] Sent: 26 April 2006 09:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] test numbers in different countries! On Wednesday 26 April 2006 07:52, Jason Frisch wrote: How about using time announments? I list of these for each country would be great! I have some test numbers on my switch in Latvia: +371 7160201 -- echo +371 7160202 -- music :) +371 7160203 -- time Do you mean something like 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 ___ --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] Excessive Asterisk delay to answer on ZAP inbound call
On Wednesday 26 April 2006 20:59, Giorgio Incantalupo wrote: Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? It's most likely waiting on callerid info. If you set usecallerid=no in your zapata.conf you should see it pick up faster, although without callerid. HTH hads -- CChheecckk yyoouurr dduupplleexx sswwiittcchh.. ___ --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] Excessive Asterisk delay to answer on ZAP inbound call
Hi Hadley, I tried usecallerid=no but unfortunately nothing changed. I used another pc with only one TDM400P because I thought I had too many TDM400P cards but I got the same behaviour. Giorgio Incantalupo Hadley Rich wrote: On Wednesday 26 April 2006 20:59, Giorgio Incantalupo wrote: Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? It's most likely waiting on callerid info. If you set usecallerid=no in your zapata.conf you should see it pick up faster, although without callerid. HTH hads ___ --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] TDM400P: flash on analog phones doesn't work
On Wed, 2006-04-26 at 14:32 +0800, Leo Ann Boon wrote: Patrick wrote: Hi, I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? [snip] Can you check your flash button interval? Some phones produce a very short pulse when you press the button, which may not be detectable by the PBX. You can adjust the flash timing in zapata.conf Thanks for the pointer. I have no idea how to measure the flash interval. With debugging turned on I see a difference when I press the flash button and do a hook flash during an active call between two analog phones. The flash button seems to generate a DTMF tone while the hook flash does not: Press flash button on analog phone1 (debug channel Zap/1-1): [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: DTMF (1) SUBCLASS: 1 (49) ] [Zap/1-1] Do hook flash on analog phone1 (debug channel Zap/1-1): -- Started three way call on channel 1 -- Started music on hold, class 'default', on channel 'Zap/2-1' [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] -- Starting simple switch on 'Zap/1-2' -- Stopped music on hold on Zap/2-1 I also tried flash=100 in zapata.conf but that made no difference. Any ideas how I can measure the flash button interval or other suggestions? Thanks and regards, Patrick ___ --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 Phones with BLA Support
I'm looking for a confirmed list of SIP phones that have support for BLA. Thank you for any info you can provide -Tim ___ --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] Sip Phones with BLA Support
Citel Handset Gateway phones support BLA (http://www.citel.com). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tim Ferguson Sent: 26 April 2006 10:51 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip Phones with BLA Support I'm looking for a confirmed list of SIP phones that have support for BLA. Thank you for any info you can provide -Tim ___ --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] do extensions must be numbers in [EMAIL PROTECTED]
Hello to all In Asterisk, SIP clients can be registered with numbers (2001, 2002, ...) or with names (manuel, maria,...). But [EMAIL PROTECTED] only allows SIP registers to be done with numbers... Is there any way of register SIP users with names and then give them a numeric alias? Thanks Joao ___ --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: Agents -- Extensions
Just incase anybody else cares in teh future, this was my solution for
number 1. ${EXTEN} is set to the agent id
exten = _2XX,2,Set(AE=${DB(/Agents/${EXTEN})})
in my case my phone was labeled 300 so AE was set to [EMAIL PROTECTED];300
--
~Shaun
Shaun [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
How can I do the following 2 things in my dialplan?
1. find out what extension a agent is assigned to by agent id.
2. find out what agent is assigned to a extension by extension id.
Anybody know how to do this? I read some where that I might have to pull
it from the db. Example code is a plus :)
--
~Shaun
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RE: [Asterisk-Users] queues that do not play music
It works very well, but has one major flaw: the calls that get to the queue will be distributed using the queue´s strategy (random for example), but the calls that goes directly to the extensions before being queued go in a static order (roundrobin without memory) and so they will overload the first person. You may want to use the Random cmd to distribute the calls if the queue is empty. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random . Make sure you increase the percentage as you go. Ex with 10 agents: First position: 1/10 chances Second : 1/9 Third : 1/8 And so on until the last agent picks up for sure if nobody else has. 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
Re: [Asterisk-Users] Trying to set up automatic announcement upon
Try using two IVRs. The 1st 'Intro' with your 'will be recorded' message has a 1 second timeout, and the only entries in it are 'i' invalid (points back to itself), and 't' timeout (points to the 2nd IVR for dialing). Date: Tue, 25 Apr 2006 21:25:53 -0600 From: Carl Youngblood [EMAIL PROTECTED] Subject: [Asterisk-Users] Trying to set up automatic announcement upon transfer for IVR in AAH 2.8 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 I am running AAH 2.8. I have an IVR for our main phone number that allows users to dial an extension directly. I would like to have a this call may be recorded announcement played before the call gets transferred. There is not a built-in option for this in the IVR web interface, but one way I can do this is to create ring groups for each user with announcements and modify the dialplan to dial the ring groups instead of the extensions. The question is, where do I do this? What part of the dialplan should I modify to make it substitute a ring group for the dialed-in extension? Sorry to post on the asterisk users list, I know AAH is not exactly related, but there is something wrong on their forum right now. I can't post there, even though I'm logged into sourceforge. Thanks, Carl -- Mike G ___ --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] Hi...Please help me
Hi,Thank you for your response. Basically, I follow "O Reilly AsteriskTFOT.pdf" book and some other eBooks. They have mentioned how to install Asterisk in server. But, they have not mentioned What I have to install in client PC's?What hardware I need?How can I take decission to buy extra hardware (like Zaptel products) OR no need of buying extra hardware? ( I will be using Asterisk for 70 PC's and a server)Is it sufficient to buy hardware for server only OR for client PC's also?How can I connect my VoIP phone to server?How can I connect hardware to server?How can I connect PSTN line to server PC?Please guide me to complete this task. Waiting for your response. Thank you.Regards,Chandra.Gonzalo Servat [EMAIL PROTECTED] wrote: On 4/24/06, Crazy Boy <[EMAIL PROTECTED]> wrote: Hi Friends,[..snip..] --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using "Asterisk" software? If It can be implemented using "Asterisk" software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware?[..snip..]It can be done with Asterisk. For the server side, you would need toinstall Asterisk on your Fedora 5 box, Zaptel and lots of Wikireading.I don't recommend using softphones for your employee PCs. It lookslike an attractive solution at first (from a cost perspective) but inreality it's not very practical (at least that was my experience).Buying 5 x 2 port ATAs will cost you around $300-$350 which is notreally expensive considering the kind of powerful PBX you will have atyour disposal. I would have suggested some Digium hardware for the FXS(extensions) but I think it will be a lot more expensive (for 10extensions) than the ATAs solution. You could also look into a channelbank, but again it will be more expensive than the 5 ATAs. As for theFXO (incoming/outgoing PSTN) I recommend buying Digium hardware(TDM400P).Hope this helps, and good luck!Regards,Gonzalo.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Blab-away for as little as 1¢/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice.___ --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] 7960G SIP Issue
Hello all, Just got hold of a CISCO 7960G. Updated the Firmware to the latest 8.2. As you all know it has 6 lines which is why i bought it. Just would like to know from you experts if this piece will connect to 6 different providers over the internet or will it only work as 6 extns with 1 provider. Im not able to get connected to more than one provider. Also i have found that the time is not working well. It runs at 1 hour ahead even if Time Zone is set correctly. Need some advice. Thanks Dan ___ --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: Intergrate Asterisk IP PBX with Legacy PBX, continuing existing funtionality of legacy pbx
We did this type of integration.
What I did to make it easier was to get new DIDs for the asterisk extensions.
(no really new, they were from a building we were
closing)
We put in a multi PRI card in the asterisk.
DID 56XX stayed in asterisk.
DID 51XX was passed to the legacy PBX via:
exten = _51XX,1,Macro(dialout-trunk,2,${EXTEN},,) ;we used to use Dial, but
have switched over to freePBX for admin and use
their built in macro.
I was able to turn on a prefix of 9 for outbound Legacy calls, so calls are
read by asterisk with just 1 set of rules.
I did have to add
exten = _956XX,1,Goto(ext-local,${EXTEN:1},1) ; to make those legacy PBX
calls get to an asterisk extension.
Before figuring out the 9 prefix on the Legacy PBX, we just did the opposite,
but it was more work because I had to add a normal
route and one without a 9 for each VOIP provider, etc.
I did have to alter the Legacy PBX to allow the 56XX number to go out a trunk.
--
--
Steven
http://www.glimasoutheast.org
Vidura Senadeera [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi All,
I would like to explain the layout that i am trying to achive. I am so
helpless on this regard.
So here is the story
This is with regard to the setup which you can find at the
Asterisk The Future of Telephony , chapter 11, page # 196-197, I am
attaching the picture for your information.
Now I am taking a challenging step to of integrate IP PBX with our
Conventional PABX system.
*Existing Setup over view*
Our existing includes traditional Pabx, E1 Line from telecom provider,
16 direct lines another telecom provider. there are around 120 extensions.
E1 Link using for DID and 16 lines using for as hunting group.
*New Integration.*
Integrate asterisk ip PBX with legacy Pabx which continues
functionality of the existing setup
I am planning to install 2 E1 cards in Asterisk box. Remove E1 link from
legacy Pabx and fix it to 1 E1 card and other E1 card will using
to connect traditional PABX.
All previous DID's which configured with the traditional PABX will be
configured in asterisk.
Actually I am not sure that i will be able to achive this migration ,
but i am trying to acomplish.
It is very much appreciate that if anyone can guide me on this regard.
Thanks Regards,
Vidura Senadeera.
Sri Lanka.
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Re: [Asterisk-Users] 7960G SIP Issue
Hello all, Just got hold of a CISCO 7960G. Updated the Firmware to the latest 8.2. As you all know it has 6 lines which is why i bought it. Just would like to know from you experts if this piece will connect to 6 different providers over the internet or will it only work as 6 extns with 1 provider. Im not able to get connected to more than one provider. I believe the phone is supposed to be able to do it, but I haven't figured out just how. I've got a few in remote locations where we had picked up a local number for line six = direct to the phone use, and I ended up proxying them through the PBX here instead, which is not really that great. So you're not crazy... I suspect it's just a little specific config magic that isn't quite set to what the phone wants as it stands. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --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] 7960G SIP Issue
Joe Greco wrote: Hello all, Just got hold of a CISCO 7960G. Updated the Firmware to the latest 8.2. As you all know it has 6 lines which is why i bought it. Just would like to know from you experts if this piece will connect to 6 different providers over the internet or will it only work as 6 extns with 1 provider. Im not able to get connected to more than one provider. I believe the phone is supposed to be able to do it, but I haven't figured out just how. I've got a few in remote locations where we had picked up a local number for line six = direct to the phone use, and I ended up proxying them through the PBX here instead, which is not really that great. In the phone config file add proxy1_address: provider 1 proxy proxy1_port: 5060 proxy1_register: 1 proxy2_address: provider 1 proxy proxy2_port: 5060 proxy2_register: 1 Repeat these definitions for each provider then add the corresponding lineX_ parameters as needed. -Steve So you're not crazy... I suspect it's just a little specific config magic that isn't quite set to what the phone wants as it stands. ... JG -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[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
[Asterisk-Users] Re: # and call speed
The # is the manual way to say that you are done dialing.
The fix depends what is adding the delay.
If it is asterisk, look for extensions like
exten = _9.,1,Macro(dialout-trunk,1,${EXTEN:1},)
This will match any number of digits after a 9, so you have to wait for the
digit timeout to see if you are done dialing.
You could use:
exten = _9XX,
exten = _91XX,
exten = _9011., ;still using any length string for international calls.
This would find a match and start dialing when either of the first two are
matched.
If it is an IP Phone causing the delay, some of them have their own dialing
rules and take in all of the digits before sending them
to asterisk, so you would have to check in the IP phone itself for that.
--
--
Steven
http://www.glimasoutheast.org
Pasqualotto Enrico [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, if I append a # after the number asterisk call more fastly, but
which step I bypass?
Can I append this in all call automaticaly? If yes, how can I do this?
- --
Pasqualotto Enrico
email: [EMAIL PROTECTED]
web: http://www.pasqualotto.org
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[Asterisk-Users] Registering to H.323 Cisco gatekeeper
I'm having trouble registering my asterisk to a cisco gatekeeper. I do not have control over the gatekeeper, and I know that it has user info defined in an LDAP. I have a user name and a password that I can use, but I can't seem to get Asterisk to register on the gatekeeper. I can't find exactly how I'm supposed to define the gatekeeper in the h323.conf file. This is the response I get in Asterisk: == Registered channel type 'H323' (The NuFone Network's Open H.323 Channel Driver) == H.323 listener started Error registering with gatekeeper ip address. Apr 26 12:37:59 ERROR[10237]: chan_h323.c:2374 load_module: Gatekeeper registration failed. The box is supposed to be a SIP/H.323 converter. -- Nils-Anders ___ --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] Splitting Zap channels into trunks?
Kerry Garrison wrote: On a TDM2400 with 3 FXO modules, is there a way to split each line into basically being its own trunk or another way to pull off the following scenerio: PBX has 12 inbound PSTN lines 1,3,5,7 are the 714 phone number hunt group 2,4,6,8 are the 888 phone number hunt group 9-12 are fax lines Customer wants outbound calls to go out in the following order: 8,7,6,5,4,3,2,1,12,10,11,9 Asterisk does not have any way to specifically define the order zap channels are used for outgoing calls (eg, 8,7,6,5...). The only reasonable way I can think of is to rearrange the actual pstn connections so as to appear in the zap channel order that you want, and then use the group= parameter on those channels. Something like: move pstn line #8 to zap channel 1 and use group=1 move pstn line #7 to zap channel 2 and use group=1 move pstn line #6 to zap channel 3 and use group=1 etc, etc. Then send your outbound calls to g1 or G1. The inbound calls on those pstn lines will still follow whatever context you define for each zap channel. R. ___ --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] astcc: need partial pin code
I have not used astcc with pin codes so far, since I set-up the phone number as card number. Some of my users want now to dial in to the system and than use their card, which is their phone number. For that I would need a way of authentication, like a pin. I want to use something like: What is your card number: user keys in the number Enter your pin:user enter a long pin Enter your destination phone number: user enters the destination phone number Is there a code snip available for that? Keyin needs always more time, we need to allow longer spaces between the digits, therefore we need to allow the # to finish the dialstring faster. I wonder if we can use one dialstring for all: cardnumber*pin*destination-number How can a user end the call and dial a new number, without hanging up? The user has usually a desk phone (=card number), and this dialin should work parallel, but of course it assumes still that only one card is in use. bye Ronald Wiplinger ___ --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] 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my *
Hello, I have 2 analogue ports on an AlcatelPBX patched to 2 FXO ports on my *@home 2.8 running on top of CentOS. Both FXO Ports are on ONEDigium card. What I would like is: If someone calls extn 281 on myAlcatel PBX it is routed through to Extn 233 on my * thruogh FXO port/module 4If someone calls extn 282 on myAlcatel PBX it is routed through to Extn 234 on my * thruogh FXO Port/module 3 I have SIP extn 233 set up.I have SIP extn 234 set up. I have one inbound route set up as any DID / any CID with a destination set to 233 (calling 281 from myAlcatel doesring on 233).I have one zap channel set up in trunks as ZAP/g0. Currentzapata-auto.config: ; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; callerid=asreceived ; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" ; channel 1, WCTDM, inactive. ; channel 2, WCTDM, inactive.signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 3 context=from-pstn group=0 channel = 3 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4context=from-pstn group=0 channel = 4 ; Span 2: ZTDUMMY/1 "ZTDUMMY/1 1" My problem is getting 234 to answer when I dial 282 from the Alcatel as I have nothing set up for it and don't seem to be able to work out how to direct different FXO ports to different * extns. I am told that my Alcatel is not passing any info (DID number etc) down the line Any help would be greatfully appreciated. Regards Andy GreenIT ManagerGBeyeLtd1 Russell StKelham IslandSheffieldS3 8RW Tel: 0114 252 1611Fax: 0114 272 9599 mailto:[EMAIL PROTECTED]http://www.businessgbeye.com Please read: CHANGE OF COMPANY NAME. As of 1st January 2006 GB Posters Ltd will be known as GB eye Ltd, please update all records and email addresses: Please replace everything after the @ in email addresses with gbeye.com (e.g. [EMAIL PROTECTED] is now [EMAIL PROTECTED]) The GB eye Ltd business website can be found at http://www.businessgbeye.com, please update your bookmarks and favourites. This e-mail is intended for the addressee(s) named above and any other use is prohibited. It may contain confidential information. If you received this e-mail in error please contact the sender by return e-mail. GB eye Ltd does not accept legal responsibility for the contents of this message if it has reached you via the Internet. Any opinions expressed are those of the author and are not necessarily endorsed by GB eye Ltd. Recipients are advised to apply their own virus checks to this message and all incoming email on delivery. ___ --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] Sphinx2
I have a gateway, which I call from my mobile phone (free of charge, since it is the same phone company). This gateway gives me a dial tone. I can than dial to any extension number or even other gateways, It is getting more a trouble to remember all the numbers, or to key in all the long phone numbers when you got the dialtone. I was thinking of using for this Sphinx2. How can I implement that? I should dial to a sphinx2 extension number, what could be 111 and than I say the name of the user I want to call to. Example: 1. dialing the gateway 2. waiting for the dialtone 3. key in 111 4. waiting for the prompt 5. say the users name: Peter 6. call Peter's number:2345678 Has anybody done something like that (partially) before? bye Ronald Wiplinger ___ --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] annoying noise on analog phones on tdm400p
Sounds like a classic case of the card being on the same IRQ as some other device in the system. cat /proc/interrupts will give you additional information. You'll have to move the card to a different slot if you find that it is sharing an IRQ. Thomas Artner wrote: hmm.. does really nobody had such an issue before? Thomas Artner wrote: Hi! I am using asterisk with two tdm400p cards. Sometimes (one call out of ten), when a call comes in and is taken, there is some terrible noise for a short time in the line (for about a second). Both partys can hear the noise. And sometimes the call has to be hung up, because the noise doesn't disappear. Has anyone any idea where the problem could be? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Another undefined pri_restart failure
Fred Noris wrote: Hi: I upgraded SuSE to 10 and Asterisk to trunk and now after deleting all modules and previously compiled stuff and recompiling asterisk, zaptel, and libpri, I get this failure of asterisk to start: [pbx_realtime.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style pbx_realtime.so (0x31) loaded RTLD_LOCAL = (Realtime Switch) [chan_mgcp.so]Apr 25 03:36:41 WARNING[8269]: loader.c:726 __load_resource: new style chan_mgcp.so (0x1) loaded RTLD_LOCAL = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found == MGCP Listening on 0.0.0.0:2727 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [chan_zap.so]Apr 25 03:36:41 WARNING[8269]: loader.c:718 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_restart Apr 25 03:36:41 WARNING[8269]: loader.c:850 print_and_load: Loading module chan_zap.so failed! I modified modules.conf to add noload = res_snmp.so, because it fails. I've tried recompiling libpri and everything and modifying path variables. Please help!! It looks like you are using Zaptel/libpri 1.0.x with Asterisk 1.2.x. Don't do that. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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 am looking for a webphone on MY SITE
I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger ___ --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] test numbers in different countries!
On Wednesday 26 April 2006 11:48, Dmitry Ivanov wrote:
On Wednesday 26 April 2006 07:52, Jason Frisch wrote:
How about using time announments? I list of these
for each country would be great!
I have some test numbers on my switch in Latvia:
+371 7160201 -- echo
+371 7160202 -- music :)
+371 7160203 -- time
+371 7160204 -- SayDigits(${CALLERIDNUM});
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Re: [Asterisk-Users] Touch tone recognition issues
Bryan Mahin wrote: I'm experiencing touch tone recognition issues when calling some outside phone systems. For instance, if I call my Nextel phone, and try to press * to enter my voicemail, Nextel's system does not hear the DTMF tone. I've also experienced other outside phone systems for which I am unable to use their touch tone menus. Oddly, this isn't the case with all outside systems. If I call Dell, everything works great. Is this a known issue with asterisk? I'm hope there is a simple setting I've over looked. This is from /path/to/src/asterisk/configs/zapata.conf.sample ; How long generated tones (DTMF and MF) will be played on the channel ; (in miliseconds) Check there to see the option you must change to increase the DTMF tone duration. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] Excessive Asterisk delay to answer on ZAP inbound call
Paste your zapata.conf. Giorgio Incantalupo wrote: Hi Hadley, I tried usecallerid=no but unfortunately nothing changed. I used another pc with only one TDM400P because I thought I had too many TDM400P cards but I got the same behaviour. Giorgio Incantalupo Hadley Rich wrote: On Wednesday 26 April 2006 20:59, Giorgio Incantalupo wrote: Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? It's most likely waiting on callerid info. If you set usecallerid=no in your zapata.conf you should see it pick up faster, although without callerid. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --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] ODBC Storage for voicemail messages in database
Seems like other postings tend to think that saving recordings as files and not as blobs in the database are a more reliable way to go. Opinions on this? Looking at supporting it for ARI and judging interest. Dan 512.791.0137 www.littlejohnconsulting.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] 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my *
well, on analog lines you generally you won't have incoming DID informations; so you'll have all calls routed to extension 's' in the context defined for that channels. By the way, you can check in the dialplan which zap channel number the call is coming in from.
To do this, you have to extract the channel number with some string manipulation:
Zap incoming channel names come in this form
Zap/channum-additionaluniqueid
So you have to CUT the channel name two times to obtain the zap channel number.
Example:
...suppose call is coming in from channel 4
[from-pstn]
;CHANNEL is Zap/4-someexten = s,1,Set(channum=${CUT(CHANNEL|-|1)}) ; channum is now Zap/4
exten = s,2,Set(channum=${CUT(channum|/|2)}) ; channum is now 4
exten = s,3,Gotoif(${channum}=3?233|1) ; go to 233 if channel is 3
exten = s,4,Gotoif(${channum}=4?234|1) ; go to 234 if channel is 4
... you can add as many as you want...
exten = s,n,Goto(233) ; go to your default extension if channel is different from
;the ones specified above
...
exten = 233,1,Dial(SIP/233)
exten = 234,1,Dial(SIP/234)
Hope this helps...
2006/4/26, Andy Green [EMAIL PROTECTED]:
Hello,
I have 2 analogue ports on an AlcatelPBX patched to 2 FXO ports on my
*@home 2.8 running on top of CentOS. Both FXO Ports are on ONEDigium card.
What I would like is:
If someone calls extn 281 on myAlcatel PBX it is routed through to Extn 233 on my * thruogh FXO port/module 4If someone calls extn 282 on myAlcatel
PBX it is routed through to Extn 234 on my * thruogh FXO Port/module 3
I have SIP extn 233 set up.I have SIP extn 234 set up.
I have one inbound route set up as any DID / any CID with a destination set to 233 (calling 281 from myAlcatel doesring on 233).
I have one zap channel set up in trunks as ZAP/g0.
Currentzapata-auto.config:
; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf)
; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ;
callerid=asreceived ; Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 ; channel 1, WCTDM, inactive.
; channel 2, WCTDM, inactive.signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 3 context=from-pstn
group=0 channel = 3 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel
4context=from-pstn group=0 channel = 4 ; Span 2: ZTDUMMY/1 ZTDUMMY/1 1
My problem is getting 234 to answer when I dial 282 from the Alcatel as I have nothing set up for it and don't seem to be able to work out how to direct different FXO ports to different
* extns.
I am told that my Alcatel is not passing any info (DID number etc) down the line
Any help would be greatfully appreciated.
Regards
Andy GreenIT ManagerGBeyeLtd
1 Russell StKelham IslandSheffieldS3 8RW
Tel: 0114 252 1611Fax: 0114 272 9599
mailto:[EMAIL PROTECTED]
http://www.businessgbeye.com
Please read: CHANGE OF COMPANY NAME. As of 1st January 2006 GB Posters Ltd will be known as GB eye Ltd, please update all records and email addresses: Please replace everything after the @ in email addresses with
gbeye.com (e.g.
[EMAIL PROTECTED] is now [EMAIL PROTECTED]) The GB eye Ltd business website can be found at
http://www.businessgbeye.com, please update your bookmarks and favourites. This e-mail is intended for the addressee(s) named above and any other use is prohibited. It may contain confidential information.
If you received this e-mail in error please contact the sender by return e-mail. GB eye Ltd does not accept legal responsibility for the contents of this message if it has reached you via the Internet. Any opinions expressed are those of the author and are not
necessarily endorsed by GB eye Ltd. Recipients are advised to apply their own virus checks to this message and all incoming email on delivery. ___
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[Asterisk-Users] IAX calls dropping after minutes
One of my PBXs drops calls after 7 to 10 minutes. I cannot see any reason for this. I upgraded to asterisk 1.2.7.1 last night, still no improvement. Calls are IAX2 to either teliax or voxee, doesn't seem to matter which. Codec is G729. Connecting over ADSL. Load is only onw or two calls, server is P4 2.4 GHz. Monitoring the ADLS does not show any significant packet loss. Watching the CLI does not show any events, the calls just end. I am at a loss, what can I do to debug this? -- Chris Mason -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Configuring QoS Params in UIP-200
I have a bunch of UIP-200 phones working in different locations. However, in one particular location the conversations sound very choppy and my client is not tolerating it. Looking through the TFTP configuration file, I see there are a bunch of parameters that could adjust the jitter and other stuff. Does anyone have any idea how to fine tune these phones for better QoS? These phones are behind a DSL line which is barely used at all, so practically, the entire DSL circuit is dedicated to these phones. Even on a single conversation where they have, at least, 128Kbps of bandwidth the conversation sounds choppy. Please help. Thanks, Waldo ___ --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] USB conference phone
I don't know about this phone but I can tell you I have a vendor that will only talk to me via Skype so I purchased this: http://www.provantage.com/usb-internet-phone~220150620.htm It operatesnice and has very good call quality. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Tuesday, April 25, 2006 8:22 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] USB conference phone Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 But looking for real world feedback. Cheers, Dean This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] billing realtime
Nick Hoffman wrote: Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 concurrent calls, how do you know to cut off the 2 calls at the 5 minute mark rather than cut off both calls after 10 minutes? That is the problem I am asking about :-) -- Jon Farmer Telford, Shropshire, UK ___ --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: SOLVED: No audio when dialing in via PRI with Q.SIG
When inserting Ringing() before MeetMe()-conference picked up the call, everything works like a charm. I guess the PRI needed to see the ringing status before the call was answered. This is however never needed when dialing a SIP-extension or similar. I have also an update considering bad PRI b-channel numbering. It seems that only my first 15 channels actually work. Then our PBX tells Asterisk it should open channel 16, when it according to Asterisk should be 17, since 16 is the D-channel. This mismatch then follows all the way up to the last channel. I've read some stuff about Q.SIG. And according to that information Q.SIG has the posibility to renumber b-channels, but Asterisk doesn't seem to care about that. I have connected our PBX to other PBX'es before, so I do know that the PRI/Q.SIG actually works with other implementations. For now I have changed chan_zap.c so that it loads the channels differently, when it configures the prioffset parameter, I just lower it by one, if it's greater than 15. This actually solved all my problems, and now both incoming and outgoing calls works just fine. I know this is not a good solutions in the long run, but it will have to do for the time being :) Mvh Peter Olsson Visionutveckling AB Tel: 0303-72 92 00 -Ursprungligt meddelande- Från: Peter Olsson Skickat: den 25 april 2006 17:41 Till: asterisk-users@lists.digium.com Ämne: Updated: No audio when dialing in via PRI with Q.SIG After lots of testing I discovered that I could get the sound to work. The only thing I had been testing was MeetMe and Voicemail. But when I dialed a SIP-phone, or routed back to other phones via the PRI interface, everything works just great! The problem only seem to occur when dialing directly into Asterisk, when Asterisk sends the audio output. I have also discovered that the PRI never seem to get the signal that the call has been connected when dialing into MeetMe, it thinks it's still in the ringing state - I've discovered this by watching TAPI events showing up on my other PBX. Is this some kinf of known bug in Asterisk? I guess it's because of this I won't get any sound on these calls When dialing to a SIP phone I get all information. If anyone have any idea, I'd appreciate it. If it helps I could also send some debug logs from ISDN. Best regards, Peter Olsson Visionutveckling AB ___ --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: billing realtime
Tony Mountifield wrote: The other situation to take account of is when the caller somehow adds to his prepaid balance while he has one or more calls in progress, in order to avoid being cut off during the call. Yes, this is a issue that needs to be considered. Also each call might be on a different cost per minute depending on the number called e.g. in the UK geographic calls are costed lower then mobile calls. The only solution I can think of at the moment is to write a daemon that uses the manager interface to hold all calls in memory and manages the current call credit available at the current time per account. If the credit expires for that account it hangs up all channels for that account. The only problem at the moment is I can't figure away to dynamically play a warning to the callers. -- Jon Farmer Telford, Shropshire, UK ___ --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] MFCR2 in Brazil, someone?
Hi Melcon, thanks for answering. That means you have a working installation? Were using a Tormenta 4 ports card. unicall.conf is not important. Im not using asterisk to test. Im using testcall, the program that comes with unicall distribution. I have done a couple of installations in Mexico with digiums card and sangoma card, but this is the first one in brazil and with tormenta card. I use testcall because is easier to detect problems. From logging the tone signaling I know the first tone is sent to the telco, but we never get a tone back, so the same first tone we sent keeps there until the 5 seconds of T1 timer expires. I have incremented the timer, but does the same, the tone keeps there until the timer expires. zaptel.conf is: span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 giving a cat to /proc/zaptel/1 show all the channels configured properly. But also shows a message like this: BPV count: 192 FAS error count: 6 Not sure what it means, i will start looking in google and the source code. Any help, hint, advice will be appreciated. On 4/25/06, Melcon Moraes [EMAIL PROTECTED] wrote: Which version of unicall and spandsp are you using? How is your zaptel.conf and unicall.conf? []'s MM -Original Message- From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 25 Apr 2006 12:45:41 -0500 Delivered: Tue, 25 Apr 2006 11:48:34 Subject:[Asterisk-Users] MFCR2 in Brazil, someone? Does anybody have a working Asterisk server with Unicall using MFCR2 in Brazil? Were having problems. It seems SPANDSP never detect the tones from the telco. Im using brazil protocol variant. Im having lots of problems to find out why spandsp seems to not detect the MF tones. We send the first digit, the telco says they receive it, and respond with the proper signal to ask for the next digit, we just never detect the tone and the T1 timer times up. Some custom logs i have put in mfcr2.c point to spandsp r2_mf_rx always returning a zero value, what seems to mean OFF TONE, because it automatically sends the code to mf_tone_off_event() but without expecting tone because it never enters to mf_tone_on_event() something like this: OUR PBX = seize TELCO = seize ACK === == First DNIS tone == here we never detect the tone from the telco the server is Linux switch-cwb.jeffnetworks.com 2.6.9-34.ELsmp #1 SMP Thu Mar 9 06:23:23 GMT 2006 x86_64 x86_64 x86_64 GNU/Linux already tried different spandsp versions without success. Thanks in advance. -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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 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=1,1145987314.216908.1433.arrino.terra.com.br,5013,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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] IAX calls dropping after minutes
Chris Mason (Lists) wrote: One of my PBXs drops calls after 7 to 10 minutes. I cannot see any reason for this. I upgraded to asterisk 1.2.7.1 last night, still no improvement. Calls are IAX2 to either teliax or voxee, doesn't seem to matter which. Codec is G729. Connecting over ADSL. Load is only onw or two calls, server is P4 2.4 GHz. Monitoring the ADLS does not show any significant packet loss. Watching the CLI does not show any events, the calls just end. I am at a loss, what can I do to debug this? If I were to debug this, I'd let ethereal sniff the packets and look at those around the dropped call to see what happened at that time. Might also consider increasing the debug level and log that logger.conf while expecting a fairly large amount of data. ___ --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] MFCR2 in Brazil, someone?
I forgot. I dont think versions matter since i have tried a lot. But current versions are the most recent in soft-switch. I think is 0.0.3 for libunicall and libmfcr and spandsp. On 4/26/06, Moises Silva [EMAIL PROTECTED] wrote: Hi Melcon, thanks for answering. That means you have a working installation? Were using a Tormenta 4 ports card. unicall.conf is not important. Im not using asterisk to test. Im using testcall, the program that comes with unicall distribution. I have done a couple of installations in Mexico with digiums card and sangoma card, but this is the first one in brazil and with tormenta card. I use testcall because is easier to detect problems. From logging the tone signaling I know the first tone is sent to the telco, but we never get a tone back, so the same first tone we sent keeps there until the 5 seconds of T1 timer expires. I have incremented the timer, but does the same, the tone keeps there until the timer expires. zaptel.conf is: span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 giving a cat to /proc/zaptel/1 show all the channels configured properly. But also shows a message like this: BPV count: 192 FAS error count: 6 Not sure what it means, i will start looking in google and the source code. Any help, hint, advice will be appreciated. On 4/25/06, Melcon Moraes [EMAIL PROTECTED] wrote: Which version of unicall and spandsp are you using? How is your zaptel.conf and unicall.conf? []'s MM -Original Message- From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 25 Apr 2006 12:45:41 -0500 Delivered: Tue, 25 Apr 2006 11:48:34 Subject:[Asterisk-Users] MFCR2 in Brazil, someone? Does anybody have a working Asterisk server with Unicall using MFCR2 in Brazil? Were having problems. It seems SPANDSP never detect the tones from the telco. Im using brazil protocol variant. Im having lots of problems to find out why spandsp seems to not detect the MF tones. We send the first digit, the telco says they receive it, and respond with the proper signal to ask for the next digit, we just never detect the tone and the T1 timer times up. Some custom logs i have put in mfcr2.c point to spandsp r2_mf_rx always returning a zero value, what seems to mean OFF TONE, because it automatically sends the code to mf_tone_off_event() but without expecting tone because it never enters to mf_tone_on_event() something like this: OUR PBX = seize TELCO = seize ACK === == First DNIS tone == here we never detect the tone from the telco the server is Linux switch-cwb.jeffnetworks.com 2.6.9-34.ELsmp #1 SMP Thu Mar 9 06:23:23 GMT 2006 x86_64 x86_64 x86_64 GNU/Linux already tried different spandsp versions without success. Thanks in advance. -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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 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=1,1145987314.216908.1433.arrino.terra.com.br,5013,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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] No Caller-ID With Cisco PAP2T-NA
Hi, I just recently started using the PAP2T-NA ATA devices, and am not getting any inbound caller-id. I did get caller-ID inbound with the Sipura SPA-2002 devices that these are modeled after. Anyone have 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
RE: [Asterisk-Users] I am looking for a webphone on MY SITE
I need the same exact thing. Our site is almost all Perl with a little PHP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, April 26, 2006 7:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITE I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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
RES: [Asterisk-Users] MFCR2 in Brazil, someone?
I am have this problem too Please, if you solve this problem send me solution I live in brazil and use CTBC TELECOM -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Moises Silva Enviada em: quarta-feira, 26 de abril de 2006 10:39 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] MFCR2 in Brazil, someone? Hi Melcon, thanks for answering. That means you have a working installation? Were using a Tormenta 4 ports card. unicall.conf is not important. Im not using asterisk to test. Im using testcall, the program that comes with unicall distribution. I have done a couple of installations in Mexico with digiums card and sangoma card, but this is the first one in brazil and with tormenta card. I use testcall because is easier to detect problems. From logging the tone signaling I know the first tone is sent to the telco, but we never get a tone back, so the same first tone we sent keeps there until the 5 seconds of T1 timer expires. I have incremented the timer, but does the same, the tone keeps there until the timer expires. zaptel.conf is: span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 giving a cat to /proc/zaptel/1 show all the channels configured properly. But also shows a message like this: BPV count: 192 FAS error count: 6 Not sure what it means, i will start looking in google and the source code. Any help, hint, advice will be appreciated. On 4/25/06, Melcon Moraes [EMAIL PROTECTED] wrote: Which version of unicall and spandsp are you using? How is your zaptel.conf and unicall.conf? []'s MM -Original Message- From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Tue, 25 Apr 2006 12:45:41 -0500 Delivered: Tue, 25 Apr 2006 11:48:34 Subject:[Asterisk-Users] MFCR2 in Brazil, someone? Does anybody have a working Asterisk server with Unicall using MFCR2 in Brazil? Were having problems. It seems SPANDSP never detect the tones from the telco. Im using brazil protocol variant. Im having lots of problems to find out why spandsp seems to not detect the MF tones. We send the first digit, the telco says they receive it, and respond with the proper signal to ask for the next digit, we just never detect the tone and the T1 timer times up. Some custom logs i have put in mfcr2.c point to spandsp r2_mf_rx always returning a zero value, what seems to mean OFF TONE, because it automatically sends the code to mf_tone_off_event() but without expecting tone because it never enters to mf_tone_on_event() something like this: OUR PBX = seize TELCO = seize ACK === == First DNIS tone == here we never detect the tone from the telco the server is Linux switch-cwb.jeffnetworks.com 2.6.9-34.ELsmp #1 SMP Thu Mar 9 06:23:23 GMT 2006 x86_64 x86_64 x86_64 GNU/Linux already tried different spandsp versions without success. Thanks in advance. -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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 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=1,114 5987314.216908.1433.arrino.terra.com.br,5013,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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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: [Serusers] Sip t38 gateway tests
Thanks for these informations I would have prefer to receive them from asterisk-users instead of serusers !! May be they are sleeping . Ok i have not installed spandsp because of i don't find some scripts like in hylafax for mail2fax fax2mail i've just patched chan_sip.c Regards Harry --- Alexandr Dubovikov [EMAIL PROTECTED] a écrit : On Wed, Apr 26, 2006 at 11:27:01AM +0200, [EMAIL PROTECTED] wrote: Are you sure spandsp has also t38 support ? I use hyfalax for faxmail support. http://www.soft-switch.org/ Work is in progress on an implementation of T.38, the real-time FAX over IP protocol. No released code is available yet. However, a number of question raised about FoIP has prompted these notes on the subject. Some initial documentation on the T.38 gateway, and termination software can be found here. The latest development version of spandsp (the 0.0.3xxx series) contain work in progress support for T.38. t38-bits.tgz, which can be found in the same directory as spandsp, contains UDPTL, TPKT, app_rxfax, app_txfax and other code needed top bring T.38 support to Asterisk. and also here: http://bugs.digium.com/view.php?id=5090 Harry --- Alexandr Dubovikov [EMAIL PROTECTED] a ?crit : On Tue, Apr 25, 2006 at 11:00:15PM +0200, [EMAIL PROTECTED] wrote: No, i ve just patched chan_sip.c Wbr, -- Alexandr Dubovikov * [EMAIL PROTECTED] RusNet * mailto:[EMAIL PROTECTED] AD1-UANIC * ICQ: 122351182 * http://www.start4.info ___ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set ___ --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: billing realtime
In article [EMAIL PROTECTED], Jon Farmer [EMAIL PROTECTED] wrote: Tony Mountifield wrote: The other situation to take account of is when the caller somehow adds to his prepaid balance while he has one or more calls in progress, in order to avoid being cut off during the call. Yes, this is a issue that needs to be considered. Also each call might be on a different cost per minute depending on the number called e.g. in the UK geographic calls are costed lower then mobile calls. The only solution I can think of at the moment is to write a daemon that uses the manager interface to hold all calls in memory and manages the current call credit available at the current time per account. If the credit expires for that account it hangs up all channels for that account. The only problem at the moment is I can't figure away to dynamically play a warning to the callers. Instead of hanging up the channel, transfer it (Action: Redirect) to an extension that does Playback(warning) followed by Hangup. You can send both caller and callee there if you use the ExtraChannel parameter to Redirect. Otherwise transferring one drops the other. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] MFCR2 in Brazil, someone?
Moises Silva wrote: Hi Melcon, thanks for answering. That means you have a working installation? Were using a Tormenta 4 ports card. unicall.conf is not important. Im not using asterisk to test. Im using testcall, the program that comes with unicall distribution. I have done a couple of installations in Mexico with digiums card and sangoma card, but this is the first one in brazil and with tormenta card. I use testcall because is easier to detect problems. From logging the tone signaling I know the first tone is sent to the telco, but we never get a tone back, so the same first tone we sent keeps there until the 5 seconds of T1 timer expires. I have incremented the timer, but does the same, the tone keeps there until the timer expires. zaptel.conf is: span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 giving a cat to /proc/zaptel/1 show all the channels configured properly. But also shows a message like this: BPV count: 192 FAS error count: 6 Not sure what it means, i will start looking in google and the source code. Any help, hint, advice will be appreciated. A number of people use my software with Tormenta 2 cards (usually the Govarian ones), so they certainly can work. BPV is bipolar violations. You often get some shown, because some typically occur as the link starts up, and settles down. However, the number should not be increasing. The FAS errors are the same. Ignore the quantity. Just check it doesn't change. Regards, 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] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)
On 4/26/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Are you in the USA or Canada? I'm in Canada. Toronto, Ontario to be exact. And it was on a local ontario Asterisk users group where other people with Sangoma A200 cards mentioned they were having the same problem.. So it could be something related to both the Sangoma A200 card and my local telco (Bell Canada). On 4/26/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: John Novack wrote: Mike Garey wrote: well, the problem isn't that the card doesn't detect a disconnect, it's that it doesn't detect it immediately (or at least within a short period). Odds are that is the telco, and not the Sangoma or Digium card. That is quite normal for a 10-30 second delay. Not all telco CO's send an immediate pulse when the caller hangs up. Is there no way to detect 5-6 seconds of silence by Asterisk? This is from /path/src/asterisk/configs/voicemail.conf.sample. Amazing how much good stuff is in that directory. Especially handy to read after a significant upgrade (i.e. 1.0.x to 1.2.x) ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=128 the maxsilence setting doesn't really help, as I believe that only limits how long we record for after voicemail has already started and silence is detected. I don't want voicemail to record anything at all if the call has been disconnected before the voicemail prompt starts. What I don't understand is that this works fine on many regular telephone answering machines, which are _much_ less advanced than asterisk. If they can do it, why can't asterisk? I mean, if I call my home line and wait for the answering machine message to kick in, then hangup right before it starts recording, I don't end up with an empty message, the system just disconnects me. I'm starting to get a lot of complaints about this, and I can totally understand where my users are coming from - it's annoying and a waste of time to have to go through 3 or 4 empty messages. I guess I just need to isolate whether this is an asterisk problem or a Sangoma problem, so I can figure out what step to take next. Thanks, 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] Camp on?
Hi all, In .nl there is a feature provided by the incumbent that I would like to implement for an internal PBX setup. The incumbents feature does the following (adopted for internal PBX use, so no external/PSTN numbers are used): 1) pick up phone and dial an internal extension 2) if other side is busy, play a message press 5 to get connected once the other side becomes available 3) press 5 on phone 4) hangup 5) wait till phone starts ringing 6) pick up phone 7) other extension is automatically dialed again and you should hear it ring I believe this is called camp on. Found some examples on voip-info.org but they assume that you do not hangup the originating phone. Anyone have an idea how to implement this feature as described above? Thanks and regards, Patrick ___ --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] ODBC Storage for voicemail messages in database
Hi Dan - Seems like other postings tend to think that saving recordings as files and not as blobs in the database are a more reliable way to go. Opinions on this? Looking at supporting it for ARI and judging interest. As far as integrity of the actual data, I think you're safe either way. In either case, the messages will get to the hard drive, they'd just be accessed by different methods. In terms of availability of that data, it may be less reliable with a database. We use ODBC message storage on all our asterisk boxes, and while it has never happened to us (knock on formica), if the database service does ever go bye bye, you will not have access to the messages until the database service is running properly once again. So, I think the real question is: How reliable is my database (and the ODBC driver)?. We're using MySQL 5.0.18, and so far it has been as reliable as the machines it is running on. - Noah ___ --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] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)
Mike Garey wrote: snip ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=128 the maxsilence setting doesn't really help, as I believe that only limits how long we record for after voicemail has already started and silence is detected. I don't want voicemail to record anything at all if the call has been disconnected before the voicemail prompt starts. Makes good sense. If there is no speech, but simply x seconds of silence, the VM messges should not even exist What I don't understand is that this works fine on many regular telephone answering machines, which are _much_ less advanced than asterisk. If they can do it, why can't asterisk? It certainly could, if someone were smart enough and willing to code it. There generally seems more interest in adding new wiz bang features rather than polish up what is already mostly working. I mean, if I call my home line and wait for the answering machine message to kick in, then hangup right before it starts recording, I don't end up with an empty message, the system just disconnects me. Many machines work this way even after they start recording, but there is nothing but silence. Returned dial tone and line noise cause this to fail, however. jOHN nOVACK ___ --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: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users] what cable to connect a legacy PBX to a TE410P ?
On 04/25/06 05:58 Sangoma Techdesk said the following: At Sangoma we do quite a lot of back-to back T1 and E1 connections. T1 is not a very fussy connection, as the baud rate is only about 750 kbps. In our experience, for error free communications you can use the following rules of thumb: Up to 50 ft: Flat patch cable Up to 500 ft: Ordinary twisted telephone cable Cat 5 may be overkill unless you are going hundreds of feet. we've faced weird intermittent problems and we suspect it's related to electrical interference caused by power cables et al in the server rack. we've seen this with both sangoma and digium cards when attempting to connect asterisk boxes to carrier E1s provided by the local operator. the cables used are normal cat5 UTP cables. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Stuck in Queues
With Asterisk 1.2.7. I have 3 queues defined, that I want to work as thus. If there are no agents logged on to that queue, the call gets passed to another queue, and if that queue has no agents logged on then on to voicemail. My queues.conf looks like this: [sales] musiconhold = default announce = queue-sales strategy = ringall wrapuptime=15 timeout = 30 maxlen = 0 announce-frequency = 90 announce-holdtime = yes monitor-format = wav monitor-join = yes leavewhenempty = yes joinempty = no member = Agent/1003 member = Agent/1004 member = Agent/1005 [tech] musiconhold = default announce = queue-tech strategy = ringall wrapuptime=5 timeout = 30 maxlen = 0 announce-frequency = 90 announce-holdtime = yes monitor-format = wav monitor-join = yes leavewhenempty = yes joinempty = no member = Agent/1001 [accounts] musiconhold = default announce = queue-accounts strategy = ringall wrapuptime=5 timeout = 30 maxlen = 0 announce-frequency = 90 announce-holdtime = yes monitor-format = wav monitor-join = yes leavewhenempty = yes joinempty = no member = Agent/1002 And the relevant part of my extensions.conf looks like this: ;sales exten = 1,1,Queue(sales) exten = 1,2,Queue(accounts) exten = 1,3,Queue(tech) exten = 1,4,Voicemail(s1012) ;accounts and billing exten = 2,1,Queue(accounts) exten = 2,2,Queue(sales) exten = 2,3,Voicemail(s1012) ;customer support exten = 3,1,Queue(tech) exten = 3,2,Queue(sales) exten = 3,3,Voicemail(s1012) When the call comes in and sales is selected, the call sits in the sales queue indefinately, even though there are no agents logged in. Am I getting something very silly wrong here? TIA for any help with 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] USB conference phone
This is an excellent USB speakerphone http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim HouserSent: Wednesday, April 26, 2006 6:26 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] USB conference phone I don't know about this phone but I can tell you I have a vendor that will only talk to me via Skype so I purchased this: http://www.provantage.com/usb-internet-phone~220150620.htm It operatesnice and has very good call quality. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean CollinsSent: Tuesday, April 25, 2006 8:22 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] USB conference phone Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 But looking for real world feedback. Cheers, Dean This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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 am looking for a webphone on MY SITE
What would AJAX have anything to do with installing a softphone on your website? I think you need to be a bit more explicit? Are you looking for something that visitors to your website can use to call you?Kudos on throwing around the buzzword, though. --TomOn 4/26/06, Jim Houser [EMAIL PROTECTED] wrote: I need the same exact thing.Our site is almost all Perl with a little PHP.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of RonaldWiplingerSent: Wednesday, April 26, 2006 7:41 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITEI am looking for a way of not to install a softphone, preferable as a linkon a web site to a webphone on MY SITE !!!Has anybody an idea for that? AJAX? byeRonald Wiplinger___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersThis e-mail and any attachments may contain confidential and privileged information.If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal.Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___--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] Excessive Asterisk delay to answer on ZAP inboundcall
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Wednesday, April 26, 2006 4:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inboundcall Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __Asterisk fxo line -| -Analog phone The analog phone rings immediately when calling, while asterisk shows the message Starting simple switch on zap... after the first ring and executes the old extension script after the second ring (for example a NoOp instruction). Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? TIA Giorgio Incantalupo ___ --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] Excessive Asterisk delay to answer on ZAP inboundcall
Hi, I have make some test. If asterisk can decode the callerid. The asterisk will answer the call after 2 rings. But when asterisk have some problem to get the callerid. Asterisk pickup the call after 3-4 rings. Regards, Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Wednesday, April 26, 2006 4:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inboundcall Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __Asterisk fxo line -| -Analog phone The analog phone rings immediately when calling, while asterisk shows the message Starting simple switch on zap... after the first ring and executes the old extension script after the second ring (for example a NoOp instruction). Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? TIA Giorgio Incantalupo ___ --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] Polycom Delay
Hi Kevin - Okay, so calls going to and from office A have no problems at all. Office B is having a bit of a delay (about 5 seconds before the CLI shows the call is even started). The odd part is, it only happens when they are making an outbound call. Incoming calls go directly to them without any problems. Both offices for external calls use our PRI we have installed and all interal are SIP. I think also internal calls are having the same problem, but that I haven't had a 100% sure answer if it is or isn't, but I know for sure the PRI calls are. First, let me say that my parents are Mercury wireless subscribers in WI and they have had some latency issues with their connection. I'm in NY and tried to set up video conferencing with them. We ran into HUGE delays that made it impossible. That being said, do you have another site (maybe your home) that can connect to office B just for some reference testing? Maybe you could just use an IAX softphone. I would venture a guess that it is the wireless connection and not the NAT or phone that is causing the problems. - Noah ___ --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] No Caller-ID With Cisco PAP2T-NA
Have you tried plugging in a different phone to the PAP2? Some times certain phones require that you up the ring voltage. On 4/26/06, Matt [EMAIL PROTECTED] wrote: Hi, I just recently started using the PAP2T-NA ATA devices, and am not getting any inbound caller-id. I did get caller-ID inbound with the Sipura SPA-2002 devices that these are modeled after. Anyone have 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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] About Softphone IAX free for Pocket PC
I can't find a pocket pc version of that on the iaxcomm website.. Only linux, Mac, Windows.. Can you send a link? This is exactly what I'm looking for!! Thanks! -Original Message- From: Robert Augustyn [mailto:[EMAIL PROTECTED] Sent: Tuesday, April 25, 2006 12:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] About Softphone IAX free for Pocket PC I use IaxComm with good results on axim x51 ___ --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] Help on chan_misdn and MSN's
I've got my MSN's going, I'll just share how I did it below: My initial assumption was wrong. I'm supposed to have one section per ISDN channel listing all the MSN's chan_misdn is responsible for. When one of those MSN's is detected chan_misdn is supposed to jump into the dialplan in the specified context at the extension specified in the MSN. What I did was fairly simple. First of all I had to set immediate=yes. Unless I had that option chan_misdn would not pick up incoming ISDN calls. With that option set to yes chan_misdn did exactly what the documentation sad: It jumped in the specified context at the s extension. So my dialplan would receive no info on the called MSN. Next I entered the directory where I had the sources for chan_misdn and griped for Starting Ast ctx. It only appears in one file. Three lines lower in the source file is a line that changes the extension to s. I simply commented out that line, rebuilt chan_misdn and voila: I've got my MSN's in the dialplan! Finally I'm not sure I found a small compatibility problem between chan_misdn and the Romanian implementation of ISDN or I simply solved a configuration problem with a huge hammer but I'm happy it works! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Cosmin Prund Sent: Tuesday, April 25, 2006 10:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help on chan_misdn and MSN's Quick question: Is there a way to distinguish between calling MSN's when using chan_misdn? More info: I've got my ISDN2 (EuroISDN) up and running here in Romania with 1 base number plus 5 MSN's. Now I want to my * to do different things when receiving a call on from different MSN's (like forwarding the call to my FAX machine or forwarding the call to my mobile). The obvious way of doing this would be to set up different sections in the misdn.conf file for the same port (I only have one port), using different settings for the msns. Unfortunately it seems that the channel driver will only remember the last section it sees for a given channel so I can only use * as the msn - and that defeats the purpose. If any other info is required I'll happily provide it. I'm not including any other info at the moment because I'm unable to filter the list myself and the list of things I've been doing today is very long (starts with downloading kernel 2.6.16.11 off kernel.org, patching for mISDN, downloading chan_misdn, compiling everything... waaay too long list, most of it irrelevant) ___ --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] AGI and incoming call
Why don't you do something like this: exten = 12345678,1,Dial(10) exten = 45874521,1,Dial(11) exten = 32544884,1,Dial(12) replace Dial(10) and so on with apppriate extension. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 26 Apr 2006 08:47:03 +0200 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AGI and incoming call Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.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 ___ --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] Stuck in Queues
My queues.conf looks like this: [sales] musiconhold = default announce = queue-sales strategy = ringall wrapuptime=15 timeout = 30 maxlen = 0 announce-frequency = 90 announce-holdtime = yes monitor-format = wav monitor-join = yes leavewhenempty = yes joinempty = no member = Agent/1003 member = Agent/1004 member = Agent/1005 snip When the call comes in and sales is selected, the call sits in the sales queue indefinately, even though there are no agents logged in. I may be wrong but this is how I understand this : since you have agent defined (and not using AddQueueMember), Asterisk see them as being part of the queue, so it is not empty. try leavewhenempty=strict and joinempty=strict see http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf for details hth ___ --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] Excessive Asterisk delay to answer on ZAP inboundcall
fax detection and callerid will slow this down. I beleive it takes 2 seconds or so for callerid to be picked up. On 4/26/06, kevin ling [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Wednesday, April 26, 2006 4:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Excessive Asterisk delay to answer on ZAP inboundcall Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __Asterisk fxo line -| -Analog phone The analog phone rings immediately when calling, while asterisk shows the message Starting simple switch on zap... after the first ring and executes the old extension script after the second ring (for example a NoOp instruction). Why does Asterisk wait for these two rings? What is it doing meanwhile? Is it possible to shorten this interval to have an immediate response? TIA Giorgio Incantalupo ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 am looking for a webphone on MY SITE
Tom Hayden wrote: What would AJAX have anything to do with installing a softphone on your website? I think you need to be a bit more explicit? Are you looking for something that visitors to your website can use to call you? Kudos on throwing around the buzzword, though. Tom, I read your words several times, but I could not figure out which program to use you are referring. Maybe you could try it simple with http://.x. Thanks! bye Ronald Wiplinger -- Tom On 4/26/06, *Jim Houser* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I need the same exact thing. Our site is almost all Perl with a little PHP. -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Ronald Wiplinger Sent: Wednesday, April 26, 2006 7:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITE I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger ___ --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] kernel - module problem
I have the following error, I guess related to astersk, in my log file: Apr 26 10:28:52debian kernel: Zapata Telephony Interface Registered on major 196Apr 26 10:28:52 debian kernel: No ISA tormenta card found at dApr 26 10:28:52 debian kernel: Zapata Telephony Interface UnloadedApr 26 10:28:52 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output errorApr 26 10:28:52 debian insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesgApr 26 10:28:52 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: insmod char-major-196 failed I'm not sureit can cause problems.. does it ? Giuseppe ___ --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 am looking for a webphone on MY SITE
The only one I have heard of is WebIAXhttp://www.voip-info.org/wiki/view/WebIAXOn 4/26/06, Tom Hayden [EMAIL PROTECTED] wrote: What would AJAX have anything to do with installing a softphone on your website? I think you need to be a bit more explicit? Are you looking for something that visitors to your website can use to call you?Kudos on throwing around the buzzword, though. --TomOn 4/26/06, Jim Houser [EMAIL PROTECTED] wrote: I need the same exact thing.Our site is almost all Perl with a little PHP.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of RonaldWiplinger Sent: Wednesday, April 26, 2006 7:41 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITEI am looking for a way of not to install a softphone, preferable as a linkon a web site to a webphone on MY SITE !!!Has anybody an idea for that? AJAX? byeRonald Wiplinger___--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 This e-mail and any attachments may contain confidential and privileged information.If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal.Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___--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-- BruceNortex Networks ___ --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 am looking for a webphone on MY SITE
Are you looking for something that visitors to your website can use to call you? This is what I'm looking for. Basically a on-screen phone with "push to talk" buttons that are directed into a department queue. I'm open to any suggestions. Thanks. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom HaydenSent: Wednesday, April 26, 2006 9:40 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] I am looking for a webphone on MY SITE What would AJAX have anything to do with installing a softphone on your website? I think you need to be a bit more explicit? Are you looking for something that visitors to your website can use to call you?Kudos on throwing around the buzzword, though. --Tom On 4/26/06, Jim Houser [EMAIL PROTECTED] wrote: I need the same exact thing.Our site is almost all Perl with a little PHP.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of RonaldWiplingerSent: Wednesday, April 26, 2006 7:41 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITEI am looking for a way of not to install a softphone, preferable as a linkon a web site to a webphone on MY SITE !!!Has anybody an idea for that? AJAX? byeRonald Wiplinger___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersThis e-mail and any attachments may contain confidential and privileged information.If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal.Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] Early media after a dial command
Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten = i,1,Playback(ss-noservice,noanswer) Exten = i,2,Congestion(15) Exten = i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice message. But the early media fails when I try to do the following: Exten = 100,1,Dial(SIP/100,15) Exten = 100,2,Playback(standby,noanswer) Exten = 100,3,Dial(SIP/[EMAIL PROTECTED],20) The PSTN caller hears the ringing for the time of the 3 priorities (20s+15s+ time of standby sound file) My guess is the cisco is receiving a 183 Ringing and generates (or the remote PSTN side generates) a ring tone until the call is answered. Is there any way to get to have early media passed once a ringing is generated? Would there be a way to have asterisk generate the ring tone as early media to the switch to the standby message in early media? Thanks for your help Benjamin ___ --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] USB conference phone
Kerry, do you actually own one? Have you used it for long? What are you using it for? (jim personally I cant see the point of using your phone when I have a very good quality headset and mic.). Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, 26 April 2006 10:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] USB conference phone This is an excellent USB speakerphone http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser Sent: Wednesday, April 26, 2006 6:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] USB conference phone I don't know about this phone but I can tell you I have a vendor that will only talk to me via Skype so I purchased this: http://www.provantage.com/usb-internet-phone~220150620.htm It operatesnice and has very good call quality. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, April 25, 2006 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] USB conference phone Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewItem Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_contenttask=viewid=39Itemid=27 But looking for real world feedback. Cheers, Dean This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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 am looking for a webphone on MY SITE
Do a google on Mexuar. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jim Houser Sent: Wednesday, 26 April 2006 9:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] I am looking for a webphone on MY SITE I need the same exact thing. Our site is almost all Perl with a little PHP. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Wednesday, April 26, 2006 7:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITE I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger ___ --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 This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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] Stuck in Queues
Time Bandit wrote: I may be wrong but this is how I understand this : since you have agent defined (and not using AddQueueMember), Asterisk see them as being part of the queue, so it is not empty. try leavewhenempty=strict and joinempty=strict see http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf for details Works perfectly 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
RE: [Asterisk-Users] Camp on?
I believe this is called camp on. Found some examples on voip-info.org but they assume that you do not hangup the originating phone. Anyone have an idea how to implement this feature as described above? When I worked at Philips there were two variants: - camp on busy - camp on no answer The second one is tricky; after the destination number has been used again, the switch will dial the originator and then the destination and connect the two legs. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] No Caller-ID With Cisco PAP2T-NA
Interestingly a different phone does work. Is it possible to fry a phone by just keeping the ring voltage up higher? (as long as you don't go above industry standard)? On 4/26/06, Tom Vile [EMAIL PROTECTED] wrote: Have you tried plugging in a different phone to the PAP2? Some times certain phones require that you up the ring voltage. On 4/26/06, Matt [EMAIL PROTECTED] wrote: Hi, I just recently started using the PAP2T-NA ATA devices, and am not getting any inbound caller-id. I did get caller-ID inbound with the Sipura SPA-2002 devices that these are modeled after. Anyone have 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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] AGI and incoming call
Hello, It's not possible, because the flat file is generated since a database, and each day, there is news customers. Best regards, Olivier S. Innocent Evil a écrit : Why don't you do something like this: exten = 12345678,1,Dial(10) exten = 45874521,1,Dial(11) exten = 32544884,1,Dial(12) replace Dial(10) and so on with apppriate extension. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 26 Apr 2006 08:47:03 +0200 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AGI and incoming call Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.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 ___ --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 -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.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] 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my *
What I would like is: If someone calls extn 281 on my Alcatel PBX it is routed through to Extn 233 on my * thruogh FXO port/module 4 If someone calls extn 282 on my Alcatel PBX it is routed through to Extn 234 on my * thruogh FXO Port/module 3 I have SIP extn 233 set up. I have SIP extn 234 set up. snip My problem is getting 234 to answer when I dial 282 from the Alcatel as I have nothing set up for it and don't seem to be able to work out how to direct different FXO ports to different * extns. The most simple way is something like this : in your zapata.conf, or in your case, zapata-auto.conf (remember that if you run genzaptelconf, this file will be overwritten and all changes will be lost), modify it to put each ZAP channel in is own context ex.: context=from-pstn-line1 group=0 channel = 3 context=from-pstn-line2 channel = 4 then, in extension.conf (or for aah, in extension_custom.conf), define those 2 context [from-pstn-line1] exten = s,1,Answer exten = s,n,Dial(Sip/233,20) exten = s,n,Voicemail(u233) [from-pstn-line2] exten = s,1,Answer exten = s,n,Dial(Sip/234,20) exten = s,n,Voicemail(u234) N.B.: this is written from the top of my head, so it may contain some errors but it gives you the direction hth ___ --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] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)
On 4/26/06, John Novack [EMAIL PROTECTED] wrote: Mike Garey wrote: snip ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=128 the maxsilence setting doesn't really help, as I believe that only limits how long we record for after voicemail has already started and silence is detected. I don't want voicemail to record anything at all if the call has been disconnected before the voicemail prompt starts. Makes good sense. If there is no speech, but simply x seconds of silence, the VM messges should not even exist ideally there should be a setting to simply remove a voicemail message that contains nothing but silence, but that's not what the maxsilence setting is for. Once you start recording a voicemail message, if you stop talking, a timer begins counting. If you don't speak for maxsilence seconds, you get disconnected. It is, however, possible to use maxsilence in conjunction with the minmessage setting to prevent empty voicemails from being left, by simply setting minmessage to be 1 second longer than the maxsilence setting. So if someone hangs up before leaving a voicemail message, but the system doesn't disconnect them immediately, the system stops the recording after maxsilence (ie 3 seconds), but if you have minmessage set to 4, asterisk will abandon the message. The problem is, if someone calls and then maybe gets sidetracked, or has to speak to someone else for a few seconds, or pick up their cell phone, in the middle of leaving a message, they'll get disconnected. So this still isn't a solution. The only thing I can think of is to run a script (specified by externnotify) after a voicemail message is left, which runs some type of audio analysis process to determine whether the sound file actually contains anything, and if not, delete it. I'm not sure what tool I can use to do the silence detection (I checked the manpage for sox, but it seems to only be able to remove silence from the beginning/end of a file, not report whether a file has silence or not). Mike What I don't understand is that this works fine on many regular telephone answering machines, which are _much_ less advanced than asterisk. If they can do it, why can't asterisk? It certainly could, if someone were smart enough and willing to code it. There generally seems more interest in adding new wiz bang features rather than polish up what is already mostly working. I mean, if I call my home line and wait for the answering machine message to kick in, then hangup right before it starts recording, I don't end up with an empty message, the system just disconnects me. Many machines work this way even after they start recording, but there is nothing but silence. Returned dial tone and line noise cause this to fail, however. jOHN nOVACK ___ --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 am looking for a webphone on MY SITE
That's basically what I'm looking for but wondered if we could do it in Perl. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce ReevesSent: Wednesday, April 26, 2006 10:07 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] I am looking for a webphone on MY SITE The only one I have heard of is WebIAXhttp://www.voip-info.org/wiki/view/WebIAX On 4/26/06, Tom Hayden [EMAIL PROTECTED] wrote: What would AJAX have anything to do with installing a softphone on your website? I think you need to be a bit more explicit? Are you looking for something that visitors to your website can use to call you?Kudos on throwing around the buzzword, though. -- Tom On 4/26/06, Jim Houser [EMAIL PROTECTED] wrote: I need the same exact thing.Our site is almost all Perl with a little PHP.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of RonaldWiplingerSent: Wednesday, April 26, 2006 7:41 AMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] I am looking for a webphone on MY SITEI am looking for a way of not to install a softphone, preferable as a linkon a web site to a webphone on MY SITE !!!Has anybody an idea for that? AJAX? byeRonald Wiplinger___--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 This e-mail and any attachments may contain confidential and privileged information.If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal.Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___--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-- BruceNortex Networks This e-mail and any attachments may contain confidential and privileged information. If you are not the intended recipient, please notify the sender, or [EMAIL PROTECTED], immediately by return e-mail and destroy any copies. Any dissemination or use of this information by a person other than the intended recipient is unauthorized and may be illegal. Unless otherwise stated, opinions expressed in this e-mail are those of the author and are not endorsed by the author's employer. ___ --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: billing realtime
Won't the called party hear the warning as well if you do that? Jon FarmerTelford, Shropshire, UK - Original Message From: Tony Mountifield [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 26 April, 2006 3:08:18 PM Subject: [Asterisk-Users] Re: billing realtime Instead of hanging up the channel, transfer it (Action: Redirect) to an extension that does Playback(warning) followed by Hangup. You can send both caller and callee there if you use the ExtraChannel parameter to Redirect. Otherwise transferring one drops the other. ___ --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
