Re: [Asterisk-Users] stop monitor on transfer
I was taking the stance that is not an issue given that he is existingly recording calls anyway. Its kinda a black or white thing.. . if your recording any calls one must assume you have the legalities sorted out Its kinda being half pregnant... So to clarify - if it is legal to execute and you have the announcements/legal opinion to support your actions, then yeah... Anyway... Thats not the point of the topic nor the original question that was asked. Martin Joseph wrote: On Mar 12, 2006, at 10:35 PM, Adrian Carter wrote: I'd teach the boss to appreciate recorded calls and just ensure they are secure. In the US I think this illegal? Aren't you supposed to have some sort of notification or beeping to indicate a recorded call to the other party? Just a thought, Marty ___ --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 -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stop monitor on transfer
BTW.. without sparking a flame war, and I have no idea how accurate the information is, but it seems that 'single party consent' applies as long as the recorded is not to be used for illegal purposes. This means only one party (in this case the business) need consent to the recordings for them to be legal. I imagine the company therefore gives itself approval to record all its calls, and thus, its all legal. *shrug* IANAL, and I dont know how accurate the information is, since it was a curiosity google search and comes from a company marketing software to accomplish it.. .. but it seems fairly factual and similar to my understanding of other jurisdictions laws From: http://www.callcorder.com/phone-recording-law.htm Consent Generally, it is legal to record any conversation where all the parties to it consent (one party consent if all parties are in a state with corresponding law). The U.S. federal law only requires one-party consent to the recording of a telephone conversation, but explicitly does not protect the taping if it is done for a criminal or tortuous purpose. Many states have similar exceptions. Martin Joseph wrote: On Mar 12, 2006, at 10:35 PM, Adrian Carter wrote: I'd teach the boss to appreciate recorded calls and just ensure they are secure. In the US I think this illegal? Aren't you supposed to have some sort of notification or beeping to indicate a recorded call to the other party? Just a thought, Marty ___ --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 -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT call recording
On Mar 13, 2006, at 12:12 AM, Adrian Carter wrote: BTW.. without sparking a flame war, and I have no idea how accurate the information is, but it seems that 'single party consent' applies as long as the recorded is not to be used for illegal purposes. This means only one party (in this case the business) need consent to the recordings for them to be legal. I imagine the company therefore gives itself approval to record all its calls, and thus, its all legal. *shrug* IANAL, and I dont know how accurate the information is, since it was a curiosity google search and comes from a company marketing software to accomplish it.. .. but it seems fairly factual and similar to my understanding of other jurisdictions laws From: http://www.callcorder.com/phone-recording-law.htm Thanks for the Off Topic info ;~) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT call recording
Heh - yeah, I was just curious myself after your remark and chased it up and thought you might find it interesting since you had raised it *shrug* Martin Joseph wrote: On Mar 13, 2006, at 12:12 AM, Adrian Carter wrote: BTW.. without sparking a flame war, and I have no idea how accurate the information is, but it seems that 'single party consent' applies as long as the recorded is not to be used for illegal purposes. This means only one party (in this case the business) need consent to the recordings for them to be legal. I imagine the company therefore gives itself approval to record all its calls, and thus, its all legal. *shrug* IANAL, and I dont know how accurate the information is, since it was a curiosity google search and comes from a company marketing software to accomplish it.. .. but it seems fairly factual and similar to my understanding of other jurisdictions laws From: http://www.callcorder.com/phone-recording-law.htm Thanks for the Off Topic info ;~) ___ --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 -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] stop monitor on transfer
Well, make them appreciate it wont work, Ive tried that and they just don't want their calls to be recorded. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Adrian Carter |Sent: Monday, March 13, 2006 12:35 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] stop monitor on transfer | |I'd teach the boss to appreciate recorded calls and just |ensure they are secure. | |I know mine actually loves that his calls are recorded - not |many people counter-claim or argue about conversations once |you can trot out them actually making the statement they claim |they never did... *shrug* | |horses for courses I guess - but other than the obvious (make |em appreciate and embrace rather than shun and dismiss) im not |sure what you could do - Maybe just running stopmonitor again |will stop the first recording ? try just calling it twice on |those calls | |Anton Krall wrote: | Guys. | | This idea has been banging my headfor days now and I feel |the need to | share with you. | | Imagine this scenario: all calls come in thru a |receptionist, asterisk | records all incoming calls, the receptionist's work is to |transfer the | calls to internal people but some of them are bosses and you |know how | bosses are, they don't want their calls to be recorded, so, I have | been trying to figure a way on how to stop monitoring / |recoring calls | once they are transferred to a bosses extension while othe transferd | to other people stay on record mode. | | Anybody has done this or know of a way? | | I tried with stopmonitor but stopmonitor will stop recording |the call | between the receptionist and the boss but once the call is |transferred | and since the initial call come thru the recepcionist, the |call stays on record. | | What do you think guys? | | ___ | --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 | | | |-- |Adrian Carter |Technical Manager |Leading Edge Internet | |Web http://www.lei.net.au http://support.lei.net.au |Direct+61 2 6163 6162 Support 1 300 662 415 |E-mail[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 | ___ --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] Multiple IAX clients behind a firewall
At the moment, I can't seem to get more than one IAX client registered behind NAT... am I correct in my above assumption or have I missed something ? I've used multiple hardware IAX phones behind NAT without a problem. ___ --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] stop monitor on transfer
Ah! In case you were wondering, We are in Mexico. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Martin Joseph |Sent: Monday, March 13, 2006 1:42 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] stop monitor on transfer | | |On Mar 12, 2006, at 10:35 PM, Adrian Carter wrote: | | I'd teach the boss to appreciate recorded calls and just ensure they | are secure. | |In the US I think this illegal? Aren't you supposed to have |some sort of notification or beeping to indicate a recorded |call to the other party? | |Just a thought, |Marty | |___ |--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] MultiBRI in Australia - found one - maybe
We went to kernel 2.6.15, and at the same time went to mISDN 0.3.0 rc25, we haven't had any lockups but users are reporting dropped calls. Unfortunately for us this means dropping chan_mISDN in favour of the Cisco router containing BRI cards and then SIP from the Cisco to Asterisk. It may still be possible to use chan_capi with the mISDN drivers for the Drayteks but for us we've run out of time which is a bit of a bummer. I believe the problem is in chan_mISDN which is admittedly still an experimental driver at this stage with release candidates every few days for the past couple weeks. I'm still interested to know how you guys get along with these adapters. As I said, I think the problem is within chan_mISDN at this stage rather than in the USB adapters, so maybe using chan_CAPI on top of mISDN hardware drivers or using chan_vISDN would be the way to go until chan_mISDN matures. Craig - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 3:16 PM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Got my 2 dreytek adapters today... Dropped them on to my test system. After wadding thru my Memory of how to setup mISDN, I had it up and running within about 2 hours. You might be receiving an email from me shortly then if I get stuck. If it wasn't for these annoying public holidays (Labour day in Victoria) mine would probably have arrived today too :) Both of them operating in ptmp with no echo cancel turned on at this stage. Seems to be happy. That's quite comforting for initial testing. Could you try some faxing? And is there any way to measure latency with some hard figures, maybe by use of a repeater? Maybe something like this: Echo measurer - BRI 1 - BRI2 - echo responder. Where the measurer dials the responder, sends out a ping, and measures the delay in the response. I find it hard to believe that any USB induced latency could be measurable in milliseconds... Will drop them onto my local production box next week and see how we go :D Let us know! Thanks James ___ --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] Cisco 7960 SIP - Displaying Time
I have had no issues with 8.2 so far! Chris - Original Message - From: Tomislav Parcina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 7:10 AM Subject: RE: [Asterisk-Users] Cisco 7960 SIP - Displaying Time -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: 9. ozujak 2006 18:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 SIP - Displaying Time This issue has been fixed in SIP firmware 7.5 Omar A. Sabek Yes, and I read that SIP 7.5 firmware have some other issues. They recommend using 7.4 firmware. I'm not sure how good in new 8.2 firmware. Tomislav ___ --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] Understanding queue timeouts + possible bug found
Hi Ian, that's good enough, but why does it ringa agent/101 two times in a row when agent/103 is logged on but unavailable? I thought it would just skip agent/103, retry 101 (once) then 102 and so on Thank you l. In data Mon, 13 Mar 2006 02:12:36 +0100, Kevin P. Fleming [EMAIL PROTECTED] ha scritto: Lenz wrote: I have added asterisks to denote a behaviour I dont understand; the extension 101 is called twice in a row if 103 is unavailable. DO you think this is a bug or there is a valid reason why * behaves like this? (I'm running 1.2.4) No, there is no bug here. In 'roundrobin' mode, the queue calls the next agent after the one it started with last time. This means that when 103 gets called (and is unavailable), the call goes 101. On the next cycle, 101 gets called, because it _started_ with 103 last time. In 'rrmemory' mode, this is different: it will start with the next agent after the last one it tried to call (not where it started). Use 'rrmemory' mode, it is really what most people are thinking of when they want 'round robin' delivery of calls. ___ --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 -- Assum est, versa et manduca. ___ --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: RE : [Asterisk-Users] Voice problem
Hi,Like you said, local connections work OK. Actually I find the problem , it was something I exclude at the beginning - the bandwidth. Some wiseguy created a 80 kbit/s upload queue.But the ISDN could also cause this problem you never know. Sorry to bother you. ___ --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] Callerid on transfer
Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of A? Regard, Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX clients behind a firewall
On 13 Mar 2006, at 05:54, Adrian Carter wrote: Hi all, I've searched the wiki, and my basic assumption at this point is to run multiple IAX clients behind NAT I need to specifically code each client to use a different port and then setup that port to be forwarded from the NAT router to their private IP address. At the moment, I can't seem to get more than one IAX client registered behind NAT... am I correct in my above assumption or have I missed something ? No, asterisk's iax treats the combination of apparent-ip-address and port as unique. So in fact you do best to turn off any port forwarding. That way your NAT device will allocate different ports for each client, sharing the same IP address. This works because IAX re-registers (or qualifies) the connection every 60 seconds, which is enough to keep the mapping in most NATing router's caches. Tim Panton [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: Multiple IAX clients behind a firewall
In two cases we've had more than one IAX client work from behind the NAT. Recently however, behind a different NAT, there was a case where only one client could connect. So maybe it depends on the router? I'm really hoping that it will be able to connect in more cases than not and am looking forward to seeing what others respond to this question. Hagen Hi all, I've searched the wiki, and my basic assumption at this point is to run multiple IAX clients behind NAT I need to specifically code each client to use a different port and then setup that port to be forwarded from the NAT router to their private IP address. At the moment, I can't seem to get more than one IAX client registered behind NAT... am I correct in my above assumption or have I missed something ___ --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] Building a small Office EPABX with VoIP GW with Asterisk
Start with AAH ram On 3/12/06, Sanjay Arora [EMAIL PROTECTED] wrote: A small office with 2/4 VoIP ports, 6 PSTN 16 Analog Extensionsrequirement and a couple of high quality IP Phones (if required to improve call quality). What are the pros cons and costs for thefollowing options:1. ATA with required VoIP port, 8 x 16 Analog EPABX with six PSTNlines plugged in 2 VoIP lines (to be selected as outward PSTN line for outgoing VoIP call). If required, a couple of IP Phones.2. ATA, Channel Bank, Analog Phone Mix. IP Phone if required.3. ATA. IP Phones and couple of analog phones from a small card, forlocations that do not need outward calling/VoIP access. With regards.Sanjay.___--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] anyway to a2billing without IVR
yes its possible , check in a2billing.conf use_dnid=YES ram On 2/24/06, Asterisk Sales [EMAIL PROTECTED] wrote: Hello list, Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk). I want to dial the destination number to the asterisk. for example: user dials, exten =_011.,1,DeadAGI(a2billing) system will connect the destination and bill them. but right now we need to enter the destinationfollowed by the IVR prompts which i dont want. Thanks in advanved if anybody can help me. best regards shaon___--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
[Asterisk-Users] G729A
Hi all, Will G729A codec exhaust the CPU power? If yes, how many concurrent sessions that P4 server board that can stand? Pls advise. Btw, if G729A has been purchased and installed, what will happen to the Asterisk Server crash say hard-disk when down or faulty, any where to do back up first such as tar commands? Any advice will be appreciated tq ___ --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: RE : [Asterisk-Users] Voice problem
On Sun, 2006-03-12 at 13:33 -0800, Gabriel Afana wrote: Andrew, From what I've read, ISDN is *not* a very good platform for VoIP because it introduces a great deal of latency and jitter. Latency will cause communication to be difficult. Jitter will cause the calls to be choppy sounding. Where did you get that idea? ISDN is a digital TDM technology and as such does not have jitter and negligible latency (read up on TDM). ISDN and VoIP don't have anything to do with each other other than that an Asterisk box might be a SIP/IAX2 -- PSTN gateway using Basic Rate of Primary Rate ISDN on the trunk side on the Asterisk box. 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] Action after _caller_ has hungup(cmd Dial 'g'-option)
Hmm, both of you recommend a solution with the dial cmd in an
agi-script, i would prefer a direct solution but i guess there is none.
thanks for your replies!
regards
chris
On Sat, 11 Mar 2006 09:54:23 +
Julian J. M. [EMAIL PROTECTED] wrote:
You can use DeadAGI.
exten = _X.,1,DeadAGI(agicall.agi,${EXTEN})
now in that AGI (pseudocode)
$exten=Get parameter 1
$dialstring=SIP/mytrunk/.$exten;
$res=$agi-dial($dialstring),
//If we used deadagi, if the _caller_ hangs up, the agi keep runing here
$chres = $agi-channel_status();
$status=$chres['data'];
Here's a list of possible return values. If $status==6, then the
_callee_ hung up.
CLI show agi channel status
Usage: CHANNEL STATUS [channelname]
Returns the status of the specified channel.
If no channel name is given the returns the status of the
current channel. Return values:
0 Channel is down and available
1 Channel is down, but reserved
2 Channel is off hook
3 Digits (or equivalent) have been dialed
4 Line is ringing
5 Remote end is ringing
6 Line is up
7 Line is busy
---
Julian J. M.
On 3/10/06, Christian B [EMAIL PROTECTED] wrote:
Hello!
There's the g-option for the Dial-cmd that allows to execute the next
extensions in the current context when the callee hangs up.
I would need the same for a call where the caller hangs up, concretely
i have to inform a agi-application of the end of a call. Does someone
know a way to do this from the dialplan?
thanks
Christian
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Re: [Asterisk-Users] RE: Multiple IAX clients behind a firewall
Thanks all for advice... I have IAX clients configured pretty standard, and they work fine connecting to the asterisk server if coming via the itnernet.. but the moment I have more than one attempted client from a remote office that gets NAT'ed into the 'main' network so to speak it doesnt work. I only see one client connect and "iax2 show peers" reflects only one client from that IP and its connected on 4569. Subsequent clients attempting to connect behind the NAT router get 'connection timedout' when trying to register. If I unregister the first client, a new one can register, but the same symptoms persist (the guy I unregistered can't re-register again ...) Its a Dlink DI714P+ router in a completly vanilla out-of-the-box setup - so if IAX 'just works' - what am I missing?? :) Hagen Rode wrote: In two cases we've had more than one IAX client work from behind the NAT. Recently however, behind a different NAT, there was a case where only one client could connect. So maybe it depends on the router? I'm really hoping that it will be able to connect in more cases than not and am looking forward to seeing what others respond to this question. Hagen Hi all, I've searched the wiki, and my basic assumption at this point is to run multiple IAX clients behind NAT I need to specifically code each client to use a different port and then setup that port to be forwarded from the NAT router to their private IP address. At the moment, I can't seem to get more than one IAX client registered behind NAT... am I correct in my above assumption or have I missed something ___ --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 -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] stop monitor on transfer
I'd teach the boss to appreciate recorded calls and just ensure they are secure. In the US I think this illegal? Aren't you supposed to have some sort of notification or beeping to indicate a recorded call to the other party? Not necessarily; there are some businesses that are required to record all conversations. One example are those involved with stock trading and the SEC regulations. Not sure what qualifies as notification. I'd suspect that appropriate wording in some privacy policy mailed to all clients might be sufficient, but that's a guess. There are a fair number of senior mgmt types that don't want to become another Enron case, and would rather not have any evidence of over-selling products, company stock, etc, for obvious reasons. Doctors even become nervous relative to recordings as a large percentage are only used for negative purposes. ___ --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] Action after _caller_ has hungup(cmd Dial 'g'-option)
Hmm, both of you recommend a solution with the dial cmd in an agi-script, i would prefer a direct solution but i guess there is none. There is - H - Allow the calling party to hang up by hitting the '*' DTMF digit. I though that your main concern was how to cachup the hangup and deal with the result of a call(see my previous email ), which is bigger pain than H. Sorry misunderstanding you. Benchev ___ --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: Multiple IAX clients behind a firewall
On 13 Mar 2006, at 11:50, Adrian Carter wrote: Thanks all for advice... I have IAX clients configured pretty standard, and they work fine connecting to the asterisk server if coming via the itnernet.. but the moment I have more than one attempted client from a remote office that gets NAT'ed into the 'main' network so to speak it doesnt work. I only see one client connect and iax2 show peers reflects only one client from that IP and its connected on 4569. Subsequent clients attempting to connect behind the NAT router get 'connection timedout' when trying to register. If I unregister the first client, a new one can register, but the same symptoms persist (the guy I unregistered can't re- register again ...) Its a Dlink DI714P+ router in a completly vanilla out-of-the-box setup - so if IAX 'just works' - what am I missing?? :) I'd guess that the router is in a 'preserve source-port' mode, my NAT'd IAX connections look like this a 192.67.4.80 (D) 255.255.255.255 1720 Unmonitored b82.163.107.203 (D) 255.255.255.255 21398 Unmonitored l 82.163.107.203 (D) 255.255.255.255 23020 Unmonitored z212.158.206.61 (D) 255.255.255.255 56598 Unmonitored Note the middle 2 are from the same address but have different port numbers. Tim Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX clients behind a firewall
Tim Panton wrote: On 13 Mar 2006, at 05:54, Adrian Carter wrote: Hi all, I've searched the wiki, and my basic assumption at this point is to run multiple IAX clients behind NAT I need to specifically code each client to use a different port and then setup that port to be forwarded from the NAT router to their private IP address. At the moment, I can't seem to get more than one IAX client registered behind NAT... am I correct in my above assumption or have I missed something ? No, asterisk's iax treats the combination of apparent-ip-address and port as unique. So in fact you do best to turn off any port forwarding. That way your NAT device will allocate different ports for each client, sharing the same IP address. This works because IAX re-registers (or qualifies) the connection every 60 seconds, which is enough to keep the mapping in most NATing router's caches. There is a known exception to that, and it relates to broken Port Address Translation (PAT) code in some cheap firewalls. If the OP follows your response and still has a problem, he should consider trying another firewall/nat-box/firmware-release. ___ --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 help implementing call center features of Asterisk
I am looking for help in implementing call center on Asterisk server. How can we implement predictive dialing? How does it communicate with a CRM system? Are there consultants who can help us setup the system? Thank you, Naren ___ --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: Multiple IAX clients behind a firewall
Tim Panton wrote: On 13 Mar 2006, at 11:50, Adrian Carter wrote: Its a Dlink DI714P+ router in a completly vanilla out-of-the-box setup - so if IAX 'just works' - what am I missing?? :) This is the bizare thing... those same clients on SIP do exactly that... but the NAT clients all end up being only on port 4569 and only allowing one connection... meanwhile I can have numerous SIP connections that look just like your IAX ones below... Same client - fires up Idefisk and uses his IAX device login and he gets register timeout after the first client connects (on port 4569). Im going to junk the DI714 tomorrow (dlink mumble grumble grrr) and replace it with a more .. ahem.. full featured router. See if that helps at all. Im just at a loss as to what is different between the SIP connections and IAX through this router... there is no discernible config 'in the way' that would cause different behavious - beyond the port numbers used. I'd guess that the router is in a 'preserve source-port' mode, my NAT'd IAX connections look like this a 192.67.4.80 (D) 255.255.255.255 1720 Unmonitored b82.163.107.203 (D) 255.255.255.255 21398 Unmonitored l 82.163.107.203 (D) 255.255.255.255 23020 Unmonitored z212.158.206.61 (D) 255.255.255.255 56598 Unmonitored Note the middle 2 are from the same address but have different port numbers. Tim Tim Panton [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 -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need help implementing call center features of Asterisk
Naren Koka wrote: I am looking for help in implementing call center on Asterisk server. How can we implement predictive dialing? How does it communicate with a CRM system? Are there consultants who can help us setup the system? Hi Naren, Best place to post requests for consultants is the asterisk-biz list. Include a short description of the need with location and languages spoken. The -biz list is for commercial discussion topics and hiring a consultant falls into that category. http://lists.digium.com/mailman/listinfo/asterisk-biz Paul ___ --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] G729A
chan (Alpha Trilogies Networks) wrote: Hi all, Will G729A codec exhaust the CPU power? If yes, how many concurrent sessions that P4 server board that can stand? Pls advise. Btw, if G729A has been purchased and installed, what will happen to the Asterisk Server crash say hard-disk when down or faulty, any where to do back up first such as tar commands? Any advice will be appreciated The amount of cpu consumed in any asterisk system is the sum of all the activities in your system, not someone else's. G729 codec cpu consumption only occurs if your system requires translation from another codec to G729 for calls or playing sounds. All g729 calls from a g729 device to another g729 device operate in a pass-through mode, and do not consume codec translation cycles. There are several good references on the wiki relative to this as well as cpu sizing, etc. Take a look. The digium g729 codec lives in /usr/lib/asterisk/modules (on a fc3 box) and will be backed up if you config your backup appropriately. Not sure what needs to be backed up for the license part of their codec since the license is based on the mac address of your nic card. Someone else will need to comment on that. (Its fairly easy to re-license it with digium anyway.) ___ --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: Multiple IAX clients behind a firewall
Adrian Carter wrote: Tim Panton wrote: On 13 Mar 2006, at 11:50, Adrian Carter wrote: Its a Dlink DI714P+ router in a completly vanilla out-of-the-box setup - so if IAX 'just works' - what am I missing?? :) This is the bizare thing... those same clients on SIP do exactly that... but the NAT clients all end up being only on port 4569 and only allowing one connection... meanwhile I can have numerous SIP connections that look just like your IAX ones below... Same client - fires up Idefisk and uses his IAX device login and he gets register timeout after the first client connects (on port 4569). Im going to junk the DI714 tomorrow (dlink mumble grumble grrr) and replace it with a more .. ahem.. full featured router. See if that helps at all. Im just at a loss as to what is different between the SIP connections and IAX through this router... there is no discernible config 'in the way' that would cause different behavious - beyond the port numbers used. You might check to see if there is newer firmware available for the dlink before junking it. Had some similar issues with an older linksys and a firmware upgrade addressed the problem. If you're really interested in what the dlink is doing, use ethereal on the inside and outside edge of the box and see what its doing. Its not that hard to figure out. ___ --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] Need help implementing call center features ofAsterisk
Try looking into the manager api. Also, there are telephony server companies out there that uses asterisk for VoIP and do all their predictive algrithm themselves. google for key word predictive dialer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Naren Koka Sent: Monday, March 13, 2006 7:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Need help implementing call center features ofAsterisk I am looking for help in implementing call center on Asterisk server. How can we implement predictive dialing? How does it communicate with a CRM system? Are there consultants who can help us setup the system? Thank you, Naren ___ --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] stop monitor on transfer
Trading desk environments are always recorded. This is for conflict resolution and there is no advice to clients. It is only used if the client claims are contrary to the trader's - therefore where a loss is concerned. Rather than test the legality, it is meant to resolve matters before they become a legal issue. The client, in some cases, is another institution with another call recorder, so it is also used to verify the traders' claims. Recording is a source of comfort. - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 15:06 Subject: Re: [Asterisk-Users] stop monitor on transfer I'd teach the boss to appreciate recorded calls and just ensure they are secure. In the US I think this illegal? Aren't you supposed to have some sort of notification or beeping to indicate a recorded call to the other party? Not necessarily; there are some businesses that are required to record all conversations. One example are those involved with stock trading and the SEC regulations. Not sure what qualifies as notification. I'd suspect that appropriate wording in some privacy policy mailed to all clients might be sufficient, but that's a guess. There are a fair number of senior mgmt types that don't want to become another Enron case, and would rather not have any evidence of over-selling products, company stock, etc, for obvious reasons. Doctors even become nervous relative to recordings as a large percentage are only used for negative purposes. ___ --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] Need help implementing call center features ofAsterisk
Hello, There are two GPL Asterisk-based outbound call center systems available, GnuDialer and VICIDIAL. You can find consultants able to install each of them on their project sites: http://www.gnudialer.org http://astguiclient.sf.net/vicidial.html MATT--- On 3/13/06, Wai Wu [EMAIL PROTECTED] wrote: Try looking into the manager api. Also, there are telephony server companies out there that uses asterisk for VoIP and do all their predictive algrithm themselves. google for key word predictive dialer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Naren Koka Sent: Monday, March 13, 2006 7:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Need help implementing call center features ofAsterisk I am looking for help in implementing call center on Asterisk server. How can we implement predictive dialing? How does it communicate with a CRM system? Are there consultants who can help us setup the system? Thank you, Naren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Scrolling messages
Several times a day I get this meesage scrolling on one of our asterisk boxes: Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to alaw Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to alaw Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to alaw Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format slin since our native format has changed to alaw At first I thought it had something to do with the phones, but we changed the phones, and we still get them. There are only two codeces running on the box, alaw and g729. This box is an external GW, it has softphones (iax and sip) connected to it as well as hosted customers using Cisco 7940s and 60s. If anyone has any ideas how to get rid of these messages, or why we are getting them and ideas would be appreciated. We have several other asterisk boxes running, and none of them have this problem. Regards to all, Joe ___ --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] transfers/parked calls + polycom 501
I am trying to get parked calls/transfers working on our polycom 501s + asterisk.The transer button on the polycom phone does not seem to transfer/park the call properly. I have to use the # - 700 to park the call.Furthermore the # - 700 only works on incomming calls. If I dial out then try to transfer, the # - 700 doesn't do anything. Thanks[meetme-ext] exten = 600,1,MeetMe(1234|Mp|98765)[extentions] include = parkedcalls include = meetme-ext exten = _10XX,1,Dial(SIP/${EXTEN},20,tT) exten = _10XX,n,Answer exten = _10XX,n,VoiceMail([EMAIL PROTECTED]) exten = _10XX,n,Hangup()[voicemail] exten = _910XX,1,Wait(1) exten = _910XX,n,VoiceMailMain(${EXTEN:[EMAIL PROTECTED])[local] include = extentions include = voicemail[incoming] ;exten =
s,1,Zapateller(nocallerid) exten = s,1,Answer exten = s,n,Wait(2) exten = s,n,Set(TIMEOUT(response)=15) exten = s,n,Background(intro) exten = s,n,WaitExten() exten = s,n,Playback(vm-goodbye) exten = s,n,Hangup() exten = 0,1,Dial(${ATTENDANT},20) exten = 0,n,Playback(vm-nobodyavail) exten = 0,n,Hangup() exten = 1,1,Directory(voicemail,extentions,f) exten = 2,1,Directory(voicemail,extentions) include = meetme-ext include = extentions exten = i,1,Playback(vm-goodbye) exten = i,2,Hangup() exten = t,1,Playback(vm-goodbye) exten = t,2,Hangup()[outbound] ignorepat = 9 include = parkedcalls exten = _9XX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1}) exten = _9XX,2,Congestion() exten = _9XX,102,Congestion() exten = _91900NXX,1,Congestion()
exten = _91976NXX,1,Congestion() exten = _91[123456789]XXNXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1}) exten = _91[123456789]XXNXX,2,Congestion() exten = _91[123456789]XXNXX,102,Congestion() exten = 9911,1,Dial(${OUTBOUNDTRUNK}/ww911) exten = 9411,1,Dial(${OUTBOUNDTRUNK}/ww411) exten = 0,1,Dial(${OUTBOUNDTRUNK}/ww0)[local-access] include = local include = outbound
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Re: [Asterisk-Users] Callerid on transfer
No, at least not yet. On 3/13/06, Ronald Voermans [EMAIL PROTECTED] wrote: Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of A? Regard, Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need help implementing call center featuresofAsterisk
It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Need help implementing call center featuresofAsterisk Hello, There are two GPL Asterisk-based outbound call center systems available, GnuDialer and VICIDIAL. You can find consultants able to install each of them on their project sites: http://www.gnudialer.org http://astguiclient.sf.net/vicidial.html MATT--- On 3/13/06, Wai Wu [EMAIL PROTECTED] wrote: Try looking into the manager api. Also, there are telephony server companies out there that uses asterisk for VoIP and do all their predictive algrithm themselves. google for key word predictive dialer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Naren Koka Sent: Monday, March 13, 2006 7:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Need help implementing call center features ofAsterisk I am looking for help in implementing call center on Asterisk server. How can we implement predictive dialing? How does it communicate with a CRM system? Are there consultants who can help us setup the system? Thank you, Naren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] echo problem + choppy sound
I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help?We are all using:Polycom 501 -- asterisk -- PSTNzapata.conf: [channels] group = 1 language=en context=incoming signalling=fxs_ks switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=10.0 txgain=10.0 channel = 1-4 Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___ --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] Callerid on transfer
ok, thank you! Regards Ronald -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens C F Verzonden: maandag 13 maart 2006 15:27 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Callerid on transfer No, at least not yet. On 3/13/06, Ronald Voermans [EMAIL PROTECTED] wrote: Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of A? Regard, Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] echo problem + choppy sound
rxgain=10.0 txgain=10.0Maybe this is a problem2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] :I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help?We are all using:Polycom 501 -- asterisk -- PSTNzapata.conf: [channels] group = 1 language=en context=incoming signalling=fxs_ks switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=10.0 txgain=10.0 channel = 1-4 Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk large scale, help needed
Hi Members, I was able to install Asterisk and configure many of it's features. Currently I am using Extensions.conf for giving all my contexts and extensions. Whenever I change my extensions or add a new context I have to reload extensions.conf and practically it is not possible reloading many times as we update or add contexts many times. Please tell me what could be the best solution to avoid all this and if possible extensions.conf itself. I came to know about scripts using AGI but I am a newbie totally and I do not have any idea using them. I have seen a article in voip-info site showing some examples on AGI and PHP. I want to do something like this: Can I write a set of rules to Asterisk say from PHP using AGI and if the first rule fails then it must go to next rule and so on? If the first rule fails then Asterisk will go to the next rule or will I receive something from Asterisk that first rule failed and then I write the second rule and so on. If I do something like this then I can have a file for every context and update it and no reloading is necessary. I checked Mysql add on for Asterisk but if I add a new context then it is not going to work without reloading. Please tell me how can I do the above with Asterisk and please suggest me if there is any good alternative for doing this? Any help will be sincerely appreciated. Thanks, Venkat. ___ --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] Need help implementing call center featuresofAsterisk
Hello, VICIDIAL allows for some integration with a third party CRM web interfaces. We use it on inbound and outbound campaigns to link with our custom CRM website. VICIDIAL has a WEB FORM feature that allows for immediate webpage popups with customer information and a custom web address that can popup a CRM customer record as soon as a call is sent to an agent. It works very well for us and is quite flexible. In our case we also have some custom perl scripts that handle the loading of data into the dialer's lists on a regular basis from the CRM database. It's all pretty simple MySQL stuff. MATT--- On 3/13/06, Wai Wu [EMAIL PROTECTED] wrote: It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Need help implementing call center featuresofAsterisk Hello, There are two GPL Asterisk-based outbound call center systems available, GnuDialer and VICIDIAL. You can find consultants able to install each of them on their project sites: http://www.gnudialer.org http://astguiclient.sf.net/vicidial.html MATT--- On 3/13/06, Wai Wu [EMAIL PROTECTED] wrote: Try looking into the manager api. Also, there are telephony server companies out there that uses asterisk for VoIP and do all their predictive algrithm themselves. google for key word predictive dialer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Naren Koka Sent: Monday, March 13, 2006 7:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Need help implementing call center features ofAsterisk I am looking for help in implementing call center on Asterisk server. How can we implement predictive dialing? How does it communicate with a CRM system? Are there consultants who can help us setup the system? Thank you, Naren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Clustering
Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though... -Original Message- From: Kristian Larsson [mailto:[EMAIL PROTECTED] Sent: Monday, March 13, 2006 12:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On Fri, Mar 10, 2006 at 08:20:27PM -0700, Douglas Garstang wrote: Kevin, From the voip wiki at http://www.voip-info.org/wiki-Asterisk+sip+regcontext: If regcontext is specified, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us Pretend we have peer 123456, then put exten = 123456,2,Dial(SIP/123456) in your extensions.conf When phone 123456 becomes available and registers to the Asterisk, the dialplan will look like: exten = 123456,1,NoOp exten = 123456,2,Dial(SIP/123456) and as you know the dialplan always begin on priority 1 so if the phone is not registered you don't automatically move to priority 2. What I'm curious to know is whether there is a way to use this with SIP RealTime... there doesn't seem to exist a setting for both regexten and regcontext. Any pointers? Kristian. What does this mean exactly? How is it used? I've read the same piece of information dozens of times over the last few months and it makes as much sense to me today, as it did back then, which is about zero. Wow... IAX can be used to share registration info? I've never seen that mentioned anywhere. After reading the patchy docs on DUNDi, I kind of got the impression that it _might_ be able to do that sort of thing, but the docs where so bad they where useless. And while we're on the discussion topic, why doesn't Digium release some docs on DUNDi? It's their baby after all. It seems to be that almost no one uses it, simply because there's no docs that explain how to do it. Alternatively, if you don't have time, can you point me to anywhere where instructions on how to use regcontent is succinctly and clearly documented and explained? Doug. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Fri 3/10/2006 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering Douglas Garstang wrote: I'd just die to see an example of that. I've never seen an example that actually works. I quite distinctly remember reading somewhere (sorry, forget where) that this command was broken. It's not broken. If you find some official documentation that says so, then it needs to be fixed. If you read it somewhere else, then that source is not something you should trust. regexten in sip.conf works just fine; it can easily be used to make an extension 'appear' and 'disappear' from the desired context based on the status of the peer's registration. If that context is then shared among the Asterisk servers (via DUNDi, IAX2 switches or some other technique), then calls to that extension will be handled by the server it registered to automatically. ___ --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 -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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] Action on phone pickup
You are on the correct path with immediate, and using the s extension. Place the phone in a context that does the following: Wait,1 Palyback(hello) DISA(contezxt for outgoing calls) This whould do what you want. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl O. Pinc Sent: Monday, March 13, 2006 1:14 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Action on phone pickup How do I get asterisk to do something when I pick up a phone? For instance, I've got a regular pots phone hooked up to a zaptel interface, and I want it to vocalize hello when I pick up the phone and then give me a dial tone, wait for digits, make a call, etc. I tried the 's' extension in extensions.conf and setting 'immediate' to yes' in zapata.conf and that didn't seem to work. (Not to mention I don't know how to get dial tone and dialing behavior back after an application executes.) Anybody got a clue they can slap me with? Thanks. Asterisk SVN-branch-1.2-r8905 Karl [EMAIL PROTECTED] Free Software: You don't pay back, you pay forward. -- Robert A. Heinlein ___ --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: transfers/parked calls + polycom 501
Howdy - The transer button on the polycom phone does not seem to transfer/park the call properly. I have to use the # - 700 to park the call. If I recall, using the Polycom transfer, you have to make sure it is done as a blind transfer. The Polycom attended transfer (default) option does not work. Furthermore the # - 700 only works on incomming calls. If I dial out then try to transfer, the # - 700 doesn't do anything. This would be a matter of flags in your dial command. the 't' option assures that the receiving leg of the call can transfer, while the 'T' option allows the caller to transfer. If you do decide to use the 'T' flag on outgoing calls, you may want to change your transfer option to something other than '#' (maybe '##' instead), otherwise people using IVR systems (for banks, calling cards, etc) will be unable to press pound without initiating a transfer. - 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] stop monitor on transfer
Setup a 'non-recording' extension for the oss and transfer the call to that one. Ie: 7123,1,StopMonitor 7123,2,Goto(123,1) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adrian Carter Sent: Monday, March 13, 2006 1:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] stop monitor on transfer I'd teach the boss to appreciate recorded calls and just ensure they are secure. I know mine actually loves that his calls are recorded - not many people counter-claim or argue about conversations once you can trot out them actually making the statement they claim they never did... *shrug* horses for courses I guess - but other than the obvious (make em appreciate and embrace rather than shun and dismiss) im not sure what you could do - Maybe just running stopmonitor again will stop the first recording ? try just calling it twice on those calls Anton Krall wrote: Guys. This idea has been banging my headfor days now and I feel the need to share with you. Imagine this scenario: all calls come in thru a receptionist, asterisk records all incoming calls, the receptionist's work is to transfer the calls to internal people but some of them are bosses and you know how bosses are, they don't want their calls to be recorded, so, I have been trying to figure a way on how to stop monitoring / recoring calls once they are transferred to a bosses extension while othe transferd to other people stay on record mode. Anybody has done this or know of a way? I tried with stopmonitor but stopmonitor will stop recording the call between the receptionist and the boss but once the call is transferred and since the initial call come thru the recepcionist, the call stays on record. What do you think guys? ___ --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 -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[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 ___ --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] Callerid on transfer
This is not posible as a'standard' does not exist for rewritingg callerID after once a call is established. We have C (in example given) hang up and than B does a blind transfer. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald VoermansSent: Monday, March 13, 2006 4:05 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Callerid on transfer Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of A? Regard, Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo problem + choppy sound
Can you explain why?Giovanni Miano [EMAIL PROTECTED] wrote:rxgain=10.0 txgain=10.0Maybe this is a problem2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] :I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help?We are all using:Polycom 501 -- asterisk -- PSTNzapata.conf: [channels] group = 1 language=en context=incoming signalling=fxs_ksswitchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yescancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=10.0 txgain=10.0 channel = 1-4Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.c om --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___ --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] Junghanns, Germany ISDN settings
Thanks for the info, I am confused still ;-) It sounds like I need NT mode -- there are NTBA boxes involved at my location... And then -- what do I do about Termination of S/T Interface ?? and Power Feeding? http://www.junghanns.net/downloads/quadbrijumpersnew.pdf That's what I'm referencing Someone feed me some tips please! - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 11, 2006 5:08 AM Subject: Re: [Asterisk-Users] Junghanns, Germany ISDN settings Hi Chris, Chris Earle (CBL) wrote: I've got a Junghanns QuadBRI card which I'm going to install on a system in Germany Anyone give me some tips on the Jumper settings? I'm guessing it's going to be NT mode with p2p? I haven't used ISDN before. I'm going to also put a Digium TDM400P card in there to plug the analog phones into. I'm just worried about the jumpers and modes. It really depends what you will be hooking up to the asterisk box. If you are connecting to a telco's S0 bus you want the card to be in TE mode (Terminal Equipment). If you are using multiple ISDN lines that are coupled together as one bundle (ask the telco) you will probably neet to configure it as p2p. If all lines are singular, use p2mp. If you will be connecting to a PBX, everything is dependant on how that PBX is configured. Best regards, Florian ___ --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 message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- 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
Re: [Asterisk-Users] Clustering
On 13/03/06, Douglas Garstang [EMAIL PROTECTED] wrote: Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though... Like Kevin already said: If that context is then shared among the Asterisk servers (via DUNDi, IAX2 switches or some other technique), then calls to that extension will be handled by the server it registered to automatically. Use an IAX2 switch for a small, known number of servers. Consider DUNDi to extend into a larger, more dynamic 'cloud'. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] misdn
Hi all, I just arrived in Italy from Cebit, qhere I spoke with digium and Beronet people. They told me to try to use the mISDN stack to drive beronet and the new upcoming digium ISDN Cards. SO I searched, find http://www.beronet.com/download/card_installation_guide.pdf, and I immediately got the error: asterisk01:~ # cd /usr/src/install-misdn/ asterisk01:/usr/src/install-misdn # make install CONFIG_SMP=y !! Disable the SMP Setting in your Kernel Config. make: *** [test_preempt] Error 1 So I discovered that mISDN does not support SMP and preempitive multitasking. but how can I disable this on my Suse Linux 10.0 box ? I found somemody saying make oldconfig, but I tryed it and rebooting the pc I had no changes. So what am I doing wrong ? thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.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
Re: [Asterisk-Users] Re: transfers/parked calls + polycom 501
On Monday 13 March 2006 10:20, Noah Miller wrote: The transer button on the polycom phone does not seem to transfer/park the call properly. I have to use the # - 700 to park the call. If I recall, using the Polycom transfer, you have to make sure it is done as a blind transfer. The Polycom attended transfer (default) option does not work. How is this configured? That is, how do I configure the Polycom's transfer button to be a blind transfer? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clustering
On Mon, Mar 13, 2006 at 08:12:40AM -0700, Douglas Garstang wrote: Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though... I'm not quite certain as I build my call routing on scripts instead of Asterisk built in commands, but I beleive Dundi should be able to help you out in situations like this. Kristian -Original Message- From: Kristian Larsson [mailto:[EMAIL PROTECTED] Sent: Monday, March 13, 2006 12:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On Fri, Mar 10, 2006 at 08:20:27PM -0700, Douglas Garstang wrote: Kevin, From the voip wiki at http://www.voip-info.org/wiki-Asterisk+sip+regcontext: If regcontext is specified, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us Pretend we have peer 123456, then put exten = 123456,2,Dial(SIP/123456) in your extensions.conf When phone 123456 becomes available and registers to the Asterisk, the dialplan will look like: exten = 123456,1,NoOp exten = 123456,2,Dial(SIP/123456) and as you know the dialplan always begin on priority 1 so if the phone is not registered you don't automatically move to priority 2. What I'm curious to know is whether there is a way to use this with SIP RealTime... there doesn't seem to exist a setting for both regexten and regcontext. Any pointers? Kristian. What does this mean exactly? How is it used? I've read the same piece of information dozens of times over the last few months and it makes as much sense to me today, as it did back then, which is about zero. Wow... IAX can be used to share registration info? I've never seen that mentioned anywhere. After reading the patchy docs on DUNDi, I kind of got the impression that it _might_ be able to do that sort of thing, but the docs where so bad they where useless. And while we're on the discussion topic, why doesn't Digium release some docs on DUNDi? It's their baby after all. It seems to be that almost no one uses it, simply because there's no docs that explain how to do it. Alternatively, if you don't have time, can you point me to anywhere where instructions on how to use regcontent is succinctly and clearly documented and explained? Doug. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Fri 3/10/2006 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering Douglas Garstang wrote: I'd just die to see an example of that. I've never seen an example that actually works. I quite distinctly remember reading somewhere (sorry, forget where) that this command was broken. It's not broken. If you find some official documentation that says so, then it needs to be fixed. If you read it somewhere else, then that source is not something you should trust. regexten in sip.conf works just fine; it can easily be used to make an extension 'appear' and 'disappear' from the desired context based on the status of the peer's registration. If that context is then shared among the Asterisk servers (via DUNDi, IAX2 switches or some other technique), then calls to that extension will be handled by the server it registered to automatically. ___ --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 -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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 -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
[Asterisk-Users] Incoming Call keeps ringing when the second call arrives
Hi, I am new to this group.I searched for my problem in the forum but could not find any solution. So here it goes: In my work place we have an asterisk box. Everything works fine except the fact that when I first call the work phone number from my cell the auto-attendend works fine but If I hang-up and call the same number again, the call keeps ringing instead of auto-attendent answering. I also watched the sequence of events from the asterisk console. Here is what's hapenning: Starting simple switch on 'Zap/3-1' Executing Answer(Zap/3-1, ) in new stack Executing ResponseTimeout(Zap/3-1, 10) in new stack Set Response Timeout to 10 Executing BackGround(Zap/3-1, mt-welcome) in new stack Playing 'mt-welcome' (language 'en') Executing Wait(Zap/3-1, 1) in new stack Executing Goto(Zap/3-1, incomming|s|3) in new stack Goto (incomming,s,3) Executing BackGround(Zap/3-1, mt-welcome) in new stack Playing 'mt-welcome' (language 'en') awn extension (incomming, s, 3) exited non-zero on 'Zap/3-1' Executing Hangup(Zap/3-1, ) in new stack pawn extension (incomming, h, 1) exited non-zero on 'Zap/3-1' Hungup 'Zap/3-1' I think I try to disconnect around the bold area when the automated message is playing. So could you be kind enough to point me to the right direction to fix this problem? Thanx in advance Deniz ___ --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] Junghanns, Germany ISDN settings
Hi Chris, Chris Earle (CBL) wrote: Thanks for the info, I am confused still ;-) It sounds like I need NT mode -- there are NTBA boxes involved at my location... No, thats the point: If your telco delivers NT boxes, your equipment must use TE mode. It's always a pair: One side does NT mode, the other TE. Termination of S/T Interface ?? Usually you don't need to bother with that, the factory setting is fine. Power Feeding? Only needed in NT mode in combination with ISDN phones that require power feeding. Doesn't seem nessecary in your case. Best regards, Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk large scale, help needed
On 3/13/06, venkat kumar [EMAIL PROTECTED] wrote: Hi Members, I was able to install Asterisk and configure many of it's features. Currently I am using Extensions.conf for giving all my contexts and extensions. Whenever I change my extensions or add a new context I have to reload extensions.conf and practically it is not possible reloading many times as we update or add contexts many times. Please tell me what could be Why is it not possible to reload? whats wrong with reloading many times? the best solution to avoid all this and if possible extensions.conf itself. I came to know about scripts using AGI but I am a newbie totally and I do not have any idea using them. I have seen a article in voip-info site showing some examples on AGI and PHP. I want to do something like this: Can I write a set of rules to Asterisk say from PHP using AGI and if the first rule fails then it must go to next rule and so on? If the first rule fails then Asterisk will go to the next rule or will I receive something from Asterisk that first rule failed and then I write the second rule and so on. If I do something like this then I can have a file for every context and update it and no reloading is necessary. I checked Mysql add on for Asterisk but if I add a new context then it is not going to work without reloading. Please tell me how can I do the above with Asterisk and please suggest me if there is any good alternative for doing this? Any help will be sincerely appreciated. Thanks, Venkat. ___ --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] stop monitor on transfer
Ive tried that via agis and that doesn't seem to work because the stopmonitor is applied to the call between the receptionist and the boss not the original call between caller and reception which is later transferred to the boss. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Alexander Lopez |Sent: Monday, March 13, 2006 9:25 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] stop monitor on transfer | |Setup a 'non-recording' extension for the oss and transfer the |call to that one. | |Ie: | |7123,1,StopMonitor |7123,2,Goto(123,1) | | | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] On Behalf Of Adrian | Carter | Sent: Monday, March 13, 2006 1:35 AM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: Re: [Asterisk-Users] stop monitor on transfer | | I'd teach the boss to appreciate recorded calls and just ensure they | are secure. | | I know mine actually loves that his calls are recorded - not many | people counter-claim or argue about conversations once you can trot | out them actually making the statement they claim they never did... | *shrug* | | horses for courses I guess - but other than the obvious (make em | appreciate and embrace rather than shun and dismiss) im not |sure what | you could do - Maybe just running stopmonitor again will stop the | first recording ? try just calling it twice on those calls | | Anton Krall wrote: | Guys. | | This idea has been banging my headfor days now and I feel | the need to | share with you. | | Imagine this scenario: all calls come in thru a | receptionist, asterisk | records all incoming calls, the receptionist's work is to | transfer the | calls to internal people but some of them are bosses and | you know how | bosses are, they don't want their calls to be recorded, so, I have | been trying to figure a way on how to stop monitoring / | recoring calls | once they are transferred to a bosses extension while othe | transferd | to other people stay on record mode. | | Anybody has done this or know of a way? | | I tried with stopmonitor but stopmonitor will stop | recording the call | between the receptionist and the boss but once the call is | transferred | and since the initial call come thru the recepcionist, the | call stays on record. | | What do you think guys? | | ___ | --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 | | | | -- | Adrian Carter | Technical Manager | Leading Edge Internet | | Webhttp://www.lei.net.au http://support.lei.net.au | Direct+61 2 6163 6162 Support 1 300 662 415 | E-mail[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 | |___ |--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] 7970 Configs
To get the the dialplan working change: dialTemplate/dialTemplateto: dialTemplatedialplan.xml/dialTemplateand place a dialplan.xml file in your tftp directory.Simple dialplan.xml file:DIALTEMPLATE  TEMPLATE MATCH="*" Timeout="15"//DIALTEMPLATEalso to activate the 7914 add:addOnModulesaddOnModule idx="1"loadInformationS00105000100/loadInformation/addOnModule!-- Uncomment if you have second 7914addOnModule idx="2"loadInformationS00105000100/loadInformation/addOnModule--/addOnModulesjust after the loadInformation tag. You will need to load the S00105000100 file toyour tftp directory.The things I cannot figure out are: 1. How to set the secret for proxy registration. 2. How to define speeddials.Thanks,/canIf I recall when we first got the CCM5 development SIP loads, I got thesame result, but it was funny that * showed the phone as not registered.It may well be the fact that I have not downloaded the released version.It may be more non-CCM friendly.I'll play with it again next week if I can borrow a 70 away from thedevelopers for a while.The only thing I do not like about the 41/61/70/71 (all the java phones)is they only allow one password for all the separate lines/proxies inSIP mode. I may play with the config to see if it will allow more.-GregBTW: If you do get it to play nice, please post the xml file for us :)On Fri, 2006-03-10 at 13:56 -0600, Aaron Daniel wrote: Awesome, that works, 'cept now the dialplan doesn't work lol. I've programmed the voicemail button in, but anything I try to dial doesn't make it past the first digit. Aaron ___ --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] Cannot load wcfxo -- Please help!
I'm afraid that I am at a loss here. I am new to Asterisk, and have successfully set up SIP. But I cannot get my FXS card working, and I'm not sure what else I can try. # modprobe wcfxo /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: init_module: No such device 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 dmesg /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: insmod /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o failed /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: insmod wcfxo failed I have a Digium quad card (Freshmaker Rev J) with and FXO daughter card (S110M Rev A) in port 3 and an FXS card (X100M Rev C) in port 4. Are these old cards? Could that be a problem? I've tried this with Fedora Core 3 and Core 4. I've tried switching ports, daughter cards, and even the entire PCI card. I cannot switch slots, since there is only one PCI slot in this mini PC. Below is everything I can think of that may be of use in diagnosing the problem. (I apologize for how long this message is as a result!) Asterisk was checked out using cvs checkout -rv1-2 asterisk zaptel libpri I notice that lsmod shows _lots_ of modules -- only few of which are really necessary. But I don't see how that would be causing this problem. Thanks for any hints you may be able to provide. = zaptel.conf unused=1,2 fxsks=3 fxoks=4 loadzone = us defaultzone=us = ztcfg -vv = Channel map: Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Default) (Slaves: 04) 2 channels configured. == lspci 00:00.0 Host bridge: Intel Corporation 82865G/PE/P DRAM Controller/Host-Hub Interface (rev 02) 00:02.0 VGA compatible controller: Intel Corporation 82865G Integrated Graphics Controller (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev c2) 00:1f.0 ISA bridge: Intel Corporation 82801EB/ER (ICH5/ICH5R) LPC Interface Bridge (rev 02) 00:1f.1 IDE interface: Intel Corporation 82801EB/ER (ICH5/ICH5R) IDE Controller (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801EB (ICH5) SATA Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801EB/ER (ICH5/ICH5R) SMBus Controller (rev 02) 01:06.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) 01:07.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 01:08.0 FireWire (IEEE 1394): VIA Technologies, Inc. IEEE 1394 Host Controller (rev 80) == /proc/interrupts = CPU0 CPU1 0: 77976 71071IO-APIC-edge timer 1: 17 1IO-APIC-edge keyboard 2: 0 0 XT-PIC cascade 5: 719593 701620 IO-APIC-level wctdm 8: 1 0IO-APIC-edge rtc 11: 1115 0 IO-APIC-level libata, eth0 12: 20 0IO-APIC-edge PS/2 Mouse 14: 4939 1286IO-APIC-edge ide0 15: 64 0IO-APIC-edge ide1 NMI: 0 0 LOC: 148957 148956 ERR: 0 MIS: 0 == lsmod = Module Size Used byNot tainted usbserial 23868 0 (autoclean) (unused) parport_pc 18884 1 (autoclean) lp 9156 0 (autoclean) parport38848 1 (autoclean) [parport_pc lp] autofs416888 0 (autoclean) (unused) audit 90872 3 8139too17704 1 mii 4088 0 [8139too] crc32 3764 0 [8139too] wcusb 19584 0 (unused) wctdm 42400 0 (unused) zaptel183712 0 [wcusb wctdm] floppy 57552 0 (autoclean) sg 37388 0 (autoclean) sr_mod 17880 0 (autoclean) microcode 6912 0 (autoclean) ide-scsi 12528 0 ide-cd 34016 0 cdrom 32896 0 [sr_mod ide-cd] keybdev 2976 0 (unused) mousedev5688 0 (unused) hid22532 0 (unused) input 6176 0 [keybdev mousedev hid] usbcore81152 1 [usbserial wcusb hid] ext3 90088 2 jbd55380 2 [ext3] ata_piix5384 0 (unused) scsi_dump_register 2368 0 [ata_piix] libata 47324 0 [ata_piix] sd_mod 14160 0 (unused) dmesg == Linux version 2.4.21-37.0.1.ELsmp ([EMAIL PROTECTED]) (gcc version 3.2.3 20030502 (Red Hat Linux 3.2.3-53)) #1 SMP Thu Jan 19 14:12:32 EST 2006 BIOS-provided physical RAM map: BIOS-e820: - 0009f400 (usable) BIOS-e820: 0009f400 - 000a (reserved) BIOS-e820:
RE: [Asterisk-Users] Clustering
Now that I've read that paragraph of Kevin's a few times, it strikes me that this is not a redundant configuration. If the call is handled by the Asterisk system where the phone registered, what happens if that system becomes available? Can another system (one that did not handle the registration) process the call? -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED] Sent: Monday, March 13, 2006 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On 13/03/06, Douglas Garstang [EMAIL PROTECTED] wrote: Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though... Like Kevin already said: If that context is then shared among the Asterisk servers (via DUNDi, IAX2 switches or some other technique), then calls to that extension will be handled by the server it registered to automatically. Use an IAX2 switch for a small, known number of servers. Consider DUNDi to extend into a larger, more dynamic 'cloud'. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Clustering
Yes, people mention DUNDi ocassionaly. It's a shame it's completely useless as their is no documentation for it. -Original Message- From: Kristian Larsson [mailto:[EMAIL PROTECTED] Sent: Monday, March 13, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On Mon, Mar 13, 2006 at 08:12:40AM -0700, Douglas Garstang wrote: Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though... I'm not quite certain as I build my call routing on scripts instead of Asterisk built in commands, but I beleive Dundi should be able to help you out in situations like this. Kristian -Original Message- From: Kristian Larsson [mailto:[EMAIL PROTECTED] Sent: Monday, March 13, 2006 12:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On Fri, Mar 10, 2006 at 08:20:27PM -0700, Douglas Garstang wrote: Kevin, From the voip wiki at http://www.voip-info.org/wiki-Asterisk+sip+regcontext: If regcontext is specified, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us Pretend we have peer 123456, then put exten = 123456,2,Dial(SIP/123456) in your extensions.conf When phone 123456 becomes available and registers to the Asterisk, the dialplan will look like: exten = 123456,1,NoOp exten = 123456,2,Dial(SIP/123456) and as you know the dialplan always begin on priority 1 so if the phone is not registered you don't automatically move to priority 2. What I'm curious to know is whether there is a way to use this with SIP RealTime... there doesn't seem to exist a setting for both regexten and regcontext. Any pointers? Kristian. What does this mean exactly? How is it used? I've read the same piece of information dozens of times over the last few months and it makes as much sense to me today, as it did back then, which is about zero. Wow... IAX can be used to share registration info? I've never seen that mentioned anywhere. After reading the patchy docs on DUNDi, I kind of got the impression that it _might_ be able to do that sort of thing, but the docs where so bad they where useless. And while we're on the discussion topic, why doesn't Digium release some docs on DUNDi? It's their baby after all. It seems to be that almost no one uses it, simply because there's no docs that explain how to do it. Alternatively, if you don't have time, can you point me to anywhere where instructions on how to use regcontent is succinctly and clearly documented and explained? Doug. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Fri 3/10/2006 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering Douglas Garstang wrote: I'd just die to see an example of that. I've never seen an example that actually works. I quite distinctly remember reading somewhere (sorry, forget where) that this command was broken. It's not broken. If you find some official documentation that says so, then it needs to be fixed. If you read it somewhere else, then that source is not something you should trust. regexten in sip.conf works just fine; it can easily be used to make an extension 'appear' and 'disappear' from the desired context based on the status of the peer's registration. If that context is then shared among the Asterisk servers (via DUNDi, IAX2 switches or some other technique), then calls to that extension will be handled by the server it registered to automatically. ___ --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 -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --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:
Re: [Asterisk-Users] Clustering
On 13/03/06, Douglas Garstang [EMAIL PROTECTED] wrote: Now that I've read that paragraph of Kevin's a few times, it strikes me that this is not a redundant configuration. If the call is handled by the Asterisk system where the phone registered, what happens if that system becomes available? Can another system (one that did not handle the registration) process the call? (Any chance you could format your emails for easier quoting? Thanks) Something like this: Server A [sip-registrations] exten = peer1,2,Dial(SIP/peer1) exten = peer2,2,Dial(SIP/peer2) include = switch-server-b [switch-server-b] switch = IAX/user:[EMAIL PROTECTED]/sip-registrations So a call arriving in context sip-registrations will hit any peer which has registered (with the regcontext trick), and fall through to the 'switch' for any which hasn't. Server B has the opposite. This won't help with a failure for an in-progress call, but should automatically distrubute calls around your peers which are registered with one server or the other. If the phones know how to re-register in the event of a server failure (and I think you said you use a SRV-based system for this), then something good should be able to happen. -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --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] Cannot load wcfxo -- Please help!
Phil Freed wrote: I'm afraid that I am at a loss here. I am new to Asterisk, and have successfully set up SIP. But I cannot get my FXS card working, and I'm not sure what else I can try. # modprobe wcfxo /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: init_module: No such device 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 dmesg /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: insmod /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o failed /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: insmod wcfxo failed I have a Digium quad card (Freshmaker Rev J) with and FXO daughter card (S110M Rev A) in port 3 and an FXS card (X100M Rev C) in port 4. Are these old cards? Could that be a problem? Snip. IIRC, wcfxo is the driver for the X100P card. The 4 port analog card's driver used to be called wcfxs. but that led to the sort of confusion you're experiencing, so it was renamed to wctdm. js ___ --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] Dazed and Confused
Interesting.. replying to my earlier post. I have a TE210P that I'm trying to get up in testing.. and am getting RED ALARMS because there are no PRIs on it.. and the server keeps throwing the error... I have a single span card elsewhere that is working fine (And has PRIs in it). Can anyone explain this? On 2/14/06, vinicius zanc [EMAIL PROTECTED] wrote: I have the same problem on the same server... But I have just 3 PCI slots and the 3 are with digium cards. One of then is a TE406P with only one link connected, so there are a lot of red alarms. I'd like to have the blue light back on my server =) .. Any one already solve this? On 11/17/05, Simone Cittadini [EMAIL PROTECTED] wrote: Matt ha scritto: Hi, Just yesterday I got an amber light on my PowerEdge 2850 saying PCI Parity Error EB113 The on-screen message says: Uhhuh. NMI received. Dazed and confused, but trying to continue You probably have a hardware problem with your RAM chips I solved it putting the digium card in another pci slot (actually the first one) I think it also happened once when the card got too much red alarms for the pri coming down from provider's side, but can't be sure as the server is in housing and I don't know the exact moment when the screen went amber ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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: Delay in ringing
Hi All, I am running [EMAIL PROTECTED] with Digium TDM400 card with FXO modules plugged to PSTN lines. I am currently experiencing a delay in ringing by around 12 seconds. Is there something I need to adjust in the dial plan for this? Regards Ash ___ --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] Diff between X100M and X100P?
I have noticed a lot of folks mentioning the x100P, and very few mentioning x100M (which is what I have). Are there important differences between them? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avaya IP Office 412
Hi! First at all, sorry for my bad english ... I m trying to connect an Avaya IP Office 412 to Asterisk using E1 I ve compiled/installed libpri - zaptel - asterisk correctly and now, im trying to get the link working. I think, first step is to have green light on the TE110P, isnt it? I setup zaptel.conf: span=1,1,0,css,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 defaultzone=es So, i think: clock will be generated by Asterisk But after making ztcfg -vv , i see that all channels are correctly setup, but running zttool, always i have RED Alarm Any idea ? Thanks you very much ___ --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] music on hold without mpg123
Hello list, after the last time that mpg123 wen ballistic on our production system, we decided to skip mp3 playback altogether and to go for raw files. After half an hour playing with mpg123 and sox parameters in order to translate a mp3 file to a wav file that can be streamed back through * with no need for an mp3 decoder, I thought I'd post the result to the list to avoid wasting time in the future: The correct paramater set seems to be: mpg123 -s --rate 44100 --mono /src/mp3/fpm-sunshine.mp3 fpm-sunshine.raw sox -r 44100 -w -s -c 1 fpm-sunshine.raw -r 8000 -c 1 fpm-sunshine.wav I have also prepared a small tutorial explaining how to setup the whole thing (you can find it at http://www.oinko.net/astrecipes/index.php?n=152 - feel free to add or modify it if you think it's necessary). Thanks l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.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
RE: [Asterisk-Users] Clustering
The phone won't be able to receive any calls nor will it be able to make any calls. However, if you somehow can get the phone to register with multiple servers, the phone can still receive calls if the primary * is unavailable. How about this. I have a few Cisco 7960s which let me specify a back up proxy address so can still make out going calls if the primary is unavailable. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Douglas Garstang Sent: Monday, March 13, 2006 11:06 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Clustering Now that I've read that paragraph of Kevin's a few times, it strikes me that this is not a redundant configuration. If the call is handled by the Asterisk system where the phone registered, what happens if that system becomes available? Can another system (one that did not handle the registration) process the call? -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED] Sent: Monday, March 13, 2006 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On 13/03/06, Douglas Garstang [EMAIL PROTECTED] wrote: Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though... Like Kevin already said: If that context is then shared among the Asterisk servers (via DUNDi, IAX2 switches or some other technique), then calls to that extension will be handled by the server it registered to automatically. Use an IAX2 switch for a small, known number of servers. Consider DUNDi to extend into a larger, more dynamic 'cloud'. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Delay in ringing
Ash Thakrar wrote: Hi All, I am running [EMAIL PROTECTED] with Digium TDM400 card with FXO modules plugged to PSTN lines. I am currently experiencing a delay in ringing by around 12 seconds. Is there something I need to adjust in the dial plan for this? That's very normal. Asterisk is waiting for the callerid info from the central office (between rings 1 and 2) before it processes the inbound call. If your pstn line does not have callerid, then modify the statements in zapata.conf to disable callerid, and it will be a little quicker. ___ --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] Analog Desktop Phone
On 3/12/06, Martin Joseph [EMAIL PROTECTED] wrote: But what the OP wanted was a sulotion that together with the SAP3000 makes for something that works even when there is a blackout, since the SPA3000 allows for failover to the FXS port from the FXO port if/when there is no power to the unit. Which makes it a very good solution when needed because of 911 reasons or the like. Actually it seems to me the Sipura 3000 is overkill in that case. There are many other ATA's that are less expensive that also have a 1 port FXS, and a PSTN failover for blackout. It seems the OP doesn't need the FXO at all? The PA168V based ATA I have does this and was a little more then half the cost of a SPA3000. Works well too. Shame on me, but I already have the SPA3000. I like it very much and it works fine. Perhaps if I need another, I will look at different products. This is for my own home, where I am keeping my POTS line, partly as a 911 solution. I have found a lot of analog desktop phones that have some of the features I want, but not all of them. The Cortelco 2200 looks like it might fit the bill. But it costs about $80. I'm not sure I want to pay that much for an analog phone that isn't wireless. Other than that, the closest I have found so far is the ATT 959: http://www.amazon.com/gp/product/B00067KETY/ref=wl_it_dp/104-2261851-2083919?%5Fencoding=UTF8colid=1SGHZOJ18P2FBcoliid=I23IRSR1SF2HPGv=glancen=172282 The problem with that one is that (as near as I can tell from the photos and the manual) it has no visual MWI. Still looking, Dave ___ --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] channel manipulation
Hi: I am working on a scenario where I need to 1) create outgoing SIP channel 2) send re-INVITE 3) bridge the outgoing channel with an incoming channel scenario: user1 and user2 are in call with each other. (end-to-end RTP traffic) (when this call was placed, sip header values were dumped in a file) user3 calls user2, asterisk follows above 3 steps to establish call between user2 and user3. (transfer user2 to the new call) Does anybody know how to create a new channel and bridge two channels manually? Thanks, Jim ___ --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] Diff between X100M and X100P?
On Mon, 2006-03-13 at 11:38 -0500, Phil Freed wrote: I have noticed a lot of folks mentioning the x100P, and very few mentioning x100M (which is what I have). Are there important differences between them? The X100P is a 1 port FXO PCI card that is now discontinued by Digium. The X100M is a 1 port FXO module for the TDM400P PCI card. -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Professional Recordings
We contractedlocal voice talents to handle IVR recordings (in male female, French English). You're right that it doesn't match the system prompts, but in some cases we want that when switching to our IVRapplication. MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zach ASent: Sunday, March 12, 2006 9:24 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Professional Recordings Allison Smith is the best. Her voice can be obtained at thevoice.digium.com. See her demos at www.theivrvoice.com and youll be impressed. Plus all the voices in asterisk are from her, and I think all the voices in an ivr system should be of the same person. If you get anybody else recorded your prompts, what will you do with the voicemail, directory and some other system prompts? Or youll need to change all the required sound files too to make all the voices consistent. Zach A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk RealTime Question, Please help
Hi all, I was about to ask this question so here's an attempt to not let it get lost in the general noise on the list! Thanks, Peter Spikings. On Mon, 2006-03-13 at 01:52 -0600, [EMAIL PROTECTED] wrote: Hi All, I was able to install Asterisk and Asterisk-addons and use them successfully. But I have a problem now, I have many contexts and it looks like Asterisk is unable to find the context given directly in Mysql DB unless I specify it in Extensions.conf to switch it to RealTime. If I add a new context in Mysql then I have to add it in Extensions.conf and reload extensions whenever I need a new context. Please tell me if there is a way to avoid all this and make Asterisk take contexts directly from Mysql without mentioning that context in Extensions.conf. If this is possible then I can make my Asterisk RealTime actually and modify contexts directly in Mysql. Thanks for you help and time, Manoj. ___ --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 message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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] Hardware timing source for MeetMe
Will the low cost X100P clones available on ebay provide a solid hardware timing source? Our experience shows that while using ztdummy with no zaptel hardware does allow MeetMe to function, we experience unacceptable levels of delay after four ot five users join the conference. With both TDM400 and Sangoma A101 hardware, we have had 20+ users with no problems. We have a pure VoIP system installed, that has nor PRI or analog lines, but does have a need for MeetMe. If a $15 card will do the trick, we would obviously rather do that than spend a couple hundred bucks for the same thing. This card would not be used for voice, just timing. Thanks, Mike Clark ___ --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] Failed installing zaptel
Group Having trouble installing zaptel. Below is my server specs Intel Motherboard D101GGC TE405P CentOS-4.2-i386 Here is the output trying to do a 'make' === make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo rm -f *.ko *.mod.c .*o.cmd rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest [EMAIL PROTECTED] zaptel]# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.ELsmp/build make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:372: error: syntax error before zone_lock /usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:372: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:372: error: initializer element is not constant /usr/src/zaptel/zaptel.c:372: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:373: error: syntax error before chan_lock /usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:373: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:373: error: initializer element is not constant /usr/src/zaptel/zaptel.c:373: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c: In function `free_tone_zone': /usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel/zaptel.c:1035: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1042: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `set_tone_zone': /usr/src/zaptel/zaptel.c:1083: warning: passing arg 1 of `_read_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1095: warning: passing arg 1 of `_read_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_reg': /usr/src/zaptel/zaptel.c:1176: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1199: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1572: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1608: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_ctl_ioctl': /usr/src/zaptel/zaptel.c:3331: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function
Re: [Asterisk-Users] Diff between X100M and X100P?
Phil Freed wrote: I have noticed a lot of folks mentioning the x100P, and very few mentioning x100M (which is what I have). Are there important differences between them? The X100P was a PCI card with a single FXO port (actually a WinModem, more or less). The X100M is a daugterboard for the TDM400P card. jd ___ --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] Considering Asterisk
We currently operate an MKC Communications Server for our small company. We have 4 offices across Canada and calls to our toll-free number are answered by our VOIP server and directed by the auto attendant in the server office to the 3 satellite offices. This system works well except we have intermittent quality of the calls, sometimes losing connection altogether, but usually garbled speech etc. The 3 satellite offices are behind firewalls on adsl or cable high speed connections. We cannot get much support from MKC and I wonder if Asterisk would be a better system for this. Is this a problem because of the wide geographic area being covered, and so more router hops? -- Thomas Johnson Pacwill Environmental 527 Beaverbrook Court, Suite 420 Fredericton NB, CANADA, E3B 1X6 Tel. 506-462-0014 Fax: 506-462-0015 Email: [EMAIL PROTECTED] Internet: http://www.pacwill.ca ___ --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] Regexten Regcontext
Hi All, I've been trying to get regexten and regcontext going for some sip peers but following the examples on the wiki is not working, as far as I can tell, nothing is happening. the phone registers, sip show peers is ok, but the NoOp priority 1 extension never gets created or added to the dialplan. Has anyone got this working? Thanks. JR JR Richardson Engineering for the Masses ___ --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] Analog Desktop Phone
You really aren't going to find an analog phone that works as well as a SIP phone for what you are trying to do. Some people suggested the GXP2000 for $85 which works ok in a home environment. It is not a top quality phone but it has all the features you want plus works very nicely with Asterisk. This same conversation is constantly going on on numerous forums. If you think about what you are trying to accomplish, it might put things into perspective. You are taking a state-of-the-art phone system flush with every business feature you may ever want and trying to install it into your home and you want to use a cheap phone on it. Things are just not designed that way. If you want to be happy with your system, not to mention putting some value on your time (and heaven help you if you have a wife that will use the system) you do NOT want to use a cheap phone on this system. At a minimum go with a Linksys SPA941 or a Snom 360. You will have either one working in a matter of minutes. If you don't put any value on your time, then keep monkeying around with a lesser solution, but the few hours you will save just dropping in a decent phone should more than make up for the extra cost. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thczv F. Thczv Sent: Monday, March 13, 2006 8:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Analog Desktop Phone On 3/12/06, Martin Joseph [EMAIL PROTECTED] wrote: But what the OP wanted was a sulotion that together with the SAP3000 makes for something that works even when there is a blackout, since the SPA3000 allows for failover to the FXS port from the FXO port if/when there is no power to the unit. Which makes it a very good solution when needed because of 911 reasons or the like. Actually it seems to me the Sipura 3000 is overkill in that case. There are many other ATA's that are less expensive that also have a 1 port FXS, and a PSTN failover for blackout. It seems the OP doesn't need the FXO at all? The PA168V based ATA I have does this and was a little more then half the cost of a SPA3000. Works well too. Shame on me, but I already have the SPA3000. I like it very much and it works fine. Perhaps if I need another, I will look at different products. This is for my own home, where I am keeping my POTS line, partly as a 911 solution. I have found a lot of analog desktop phones that have some of the features I want, but not all of them. The Cortelco 2200 looks like it might fit the bill. But it costs about $80. I'm not sure I want to pay that much for an analog phone that isn't wireless. Other than that, the closest I have found so far is the ATT 959: http://www.amazon.com/gp/product/B00067KETY/ref=wl_it_dp/104-2 261851-2083919?%5Fencoding=UTF8colid=1SGHZOJ18P2FBcoliid=I23 IRSR1SF2HPGv=glancen=172282 The problem with that one is that (as near as I can tell from the photos and the manual) it has no visual MWI. Still looking, Dave ___ --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] Zaptel not compiling on lastest Centos 4.2 kernel.
Hi, I made a big mistake on a Centos 4.2 box - I forgot to exclude the kernel from updating. Now zaptel will not do a make linux26 see below. Is there a way to roll this back or is there a patch to get Zaptel to compile? I have a link to the modules using 'ln -s /lib/modules/uname -r/build linux-2.6 so that I did not have to specifiy the kernel version directly. cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.ELsmp/build make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /usr/src/zaptel-1.2.4/zaptel.o /usr/src/zaptel-1.2.4/zaptel.c:384: error: syntax error before zone_lock /usr/src/zaptel-1.2.4/zaptel.c:384: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel-1.2.4/zaptel.c:384: error: incompatible types in initialization /usr/src/zaptel-1.2.4/zaptel.c:384: error: initializer element is not constant /usr/src/zaptel-1.2.4/zaptel.c:384: warning: data definition has no type or storage class /usr/src/zaptel-1.2.4/zaptel.c:385: error: syntax error before chan_lock /usr/src/zaptel-1.2.4/zaptel.c:385: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel-1.2.4/zaptel.c:385: error: incompatible types in initialization /usr/src/zaptel-1.2.4/zaptel.c:385: error: initializer element is not constant /usr/src/zaptel-1.2.4/zaptel.c:385: warning: data definition has no type or storage class /usr/src/zaptel-1.2.4/zaptel.c: In function `free_tone_zone': /usr/src/zaptel-1.2.4/zaptel.c:1034: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:1037: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel-1.2.4/zaptel.c:1047: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:1054: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `set_tone_zone': /usr/src/zaptel-1.2.4/zaptel.c:1095: warning: passing arg 1 of `_read_lock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:1107: warning: passing arg 1 of `_read_unlock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `zt_chan_reg': /usr/src/zaptel-1.2.4/zaptel.c:1188: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:1211: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel-1.2.4/zaptel.c:1584: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:1620: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `zt_ctl_ioctl': /usr/src/zaptel-1.2.4/zaptel.c:3343: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:3345: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `zt_init': /usr/src/zaptel-1.2.4/zaptel.c:6553: error: incompatible types in assignment /usr/src/zaptel-1.2.4/zaptel.c: At top level: /usr/src/zaptel-1.2.4/zaptel.c:188: warning: 'fcstab' defined but not used make[2]: *** [/usr/src/zaptel-1.2.4/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel-1.2.4] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' make: *** [linux26] Error 2 [EMAIL PROTECTED] zaptel-1.2.4]# Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: transfers/parked calls + polycom 501
Hi Andrew - On Monday 13 March 2006 10:20, Noah Miller wrote: The transer button on the polycom phone does not seem to transfer/park the call properly. I have to use the # - 700 to park the call. If I recall, using the Polycom transfer, you have to make sure it is done as a blind transfer. The Polycom attended transfer (default) option does not work. How is this configured? That is, how do I configure the Polycom's transfer button to be a blind transfer? From what I know, you can't configure the polycom transfer button to do blind transfers by default. You just have to make sure to manually press the blind softkey every time you do a transfer for the parking lot. My solution was to set '#' as the asterisk transfer key, and remap the Polycom transfer key to '#'. Actually, my even more simplified solution was to hack parking as a feature in features.conf. I then set the '*' key to use the parking feature, and remapped the services key to '*'. I have a patch for this, if you want it. Now I need to do my part and test out the new metermaid feature in Olle's test-this-branch ;-) - 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] Hardware timing source for MeetMe
Mike Clark wrote: Will the low cost X100P clones available on ebay provide a solid hardware timing source? Our experience shows that while using ztdummy with no zaptel hardware does allow MeetMe to function, we experience unacceptable levels of delay after four ot five users join the conference. With both TDM400 and Sangoma A101 hardware, we have had 20+ users with no problems. We have a pure VoIP system installed, that has nor PRI or analog lines, but does have a need for MeetMe. If a $15 card will do the trick, we would obviously rather do that than spend a couple hundred bucks for the same thing. This card would not be used for voice, just timing. Thanks, Mike Clark ___ --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 Those do not have timing interfaces on them that I am aware of. ___ --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: Asterisk large scale, help needed
Hi - I was able to install Asterisk and configure many of it's features. Currently I am using Extensions.conf for giving all my contexts and extensions. Whenever I change my extensions or add a new context I have to reload extensions.conf and practically it is not possible reloading many times as we update or add contexts many times. Please tell me what could be Why is it not possible to reload? whats wrong with reloading many times? I think maybe there's some confusion here with the OP. Reloading does not interrupt calls in progress. They will keep on going through as many reloads as you want. the best solution to avoid all this and if possible extensions.conf itself. I came to know about scripts using AGI but I am a newbie totally and I do not have any idea using them. I have seen a article in voip-info site showing some examples on AGI and PHP. I want to do something like this: Actually, if you're talking about a large scale deployment, AGI scripts could conceivably be very bad. Depending on how they are implemented, they may add considerable processing overhead, which would be compounded on a heavily taxed server. Realtime is probably your best bet. - 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] Re: transfers/parked calls + polycom 501
When you hit the polycom's transfer button, a softkey appears on the screen that says Blind -- hitting this changes the transfer from attended to blind, and the blind button then disappears to show this. There's no real way I know to make this permanent. Andrew Kohlsmith wrote: On Monday 13 March 2006 10:20, Noah Miller wrote: The transer button on the polycom phone does not seem to transfer/park the call properly. I have to use the # - 700 to park the call. If I recall, using the Polycom transfer, you have to make sure it is done as a blind transfer. The Polycom attended transfer (default) option does not work. How is this configured? That is, how do I configure the Polycom's transfer button to be a blind transfer? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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] Failed installing zaptel
Hi, I am having the same exact problem. I am assuming that it was a problem with a kernel update I did. I am in the process of rolling back to an older kernel... I will let you let know if this works. There is also a patch for zaptel but I believe this is for going from 1.3 to 1.4? Thanks Hall, Eric M. wrote: Group Having trouble installing zaptel. Below is my server specs Intel Motherboard D101GGC TE405P CentOS-4.2-i386 Here is the output trying to do a 'make' === make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo rm -f *.ko *.mod.c .*o.cmd rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest [EMAIL PROTECTED] zaptel]# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.ELsmp/build make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:372: error: syntax error before zone_lock /usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:372: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:372: error: initializer element is not constant /usr/src/zaptel/zaptel.c:372: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:373: error: syntax error before chan_lock /usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:373: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:373: error: initializer element is not constant /usr/src/zaptel/zaptel.c:373: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c: In function `free_tone_zone': /usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel/zaptel.c:1035: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1042: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `set_tone_zone': /usr/src/zaptel/zaptel.c:1083: warning: passing arg 1 of `_read_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1095: warning: passing arg 1 of `_read_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_reg': /usr/src/zaptel/zaptel.c:1176: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1199: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1572: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1608: warning: passing arg 1 of `_write_unlock_irqrestore' from
Re: [Asterisk-Users] Professional Recordings
While I like Allison... sometimes she just sounds a little too breathy for my liking. On 3/12/06, Zach A [EMAIL PROTECTED] wrote: Allison Smith is the best. Her voice can be obtained at thevoice.digium.com. See her demos at www.theivrvoice.com and you'll be impressed. Plus all the voices in asterisk are from her, and I think all the voices in an ivr system should be of the same person. If you get anybody else recorded your prompts, what will you do with the voicemail, directory and some other system prompts? Or you'll need to change all the required sound files too to make all the voices consistent. Zach A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog Desktop Phone
On Mon, 13 Mar 2006, Kerry Garrison wrote: system) you do NOT want to use a cheap phone on this system. At a minimum go with a Linksys SPA941 or a Snom 360. You will have either one working in a I would wait until snom fixes the issues with the 360 firmware. -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
Re: [Asterisk-Users] Problem compiling zaptel on latest RHEL kernel(2.6.9-34.EL)
Thanks Russ. I updated the Makefile under /usr/src/zapteland issued rebuild_zaptel. it worked flawlessly:) On 3/13/06, Anthony Rodgers [EMAIL PROTECTED] wrote: Many thanks, Russ - I'll give this a try.Thank goodness a) for test servers and b) for the ability of Linux to rollback with a simple change to grub.conf :-)Regards,--Anthony RodgersBusiness Systems AnalystDistrict of North VancouverWeb: http://www.dnv.orgRSS Feed: http://www.dnv.org/rss.aspOn 11-Mar-06, at 7:33 AM, Russ Price wrote: Anthony Rodgers wrote: Greetings, I have just updated our test server to 2.6.9-34.EL and get the following error messages when compiling zaptel: make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M]/usr/src/zaptel/zaptel- 1.2.1/zaptel.o /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: error: syntax error before zone_lock [snipped] This bit me with CentOS 4.2 as well.The problem is actually a typo in the kernel spinlock.h file. See: http://bugs.digium.com/view.php?id=6425 and https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568 for more information. Here's a quick fix.In your zaptel Makefile, add the following (line 38 for 1.2.4) - THIS SHOLD BE ALL ONE LINE: CFLAGS+=$(shell if uname -r | grep -q 2.6.9-34.EL; then echo -Drw_lock_t=\rwlock_t\; fi) This way, if this is fixed in the next kernel release, you won't need to make another change to the Makefile. Russ ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to connect 3 or more servers via IAX ?
Hi Jean-Louis, We have 3 servers connected togther - we do it by creating specific trunks between each one. ### iax.conf from asterix server: ; IAX Trunks [dogmatix-in] type=user auth=md5 host=voip.dogmatix.dnv.org secret= context=international trunk=yes [dogmatix-out] type=peer auth=md5 host=voip.dogmatix.dnv.org username=asterix-in secret= context=international trunk=yes [obelix-in] type=user auth=md5 host=voip.obelix.dnv.org secret= context=international trunk=yes [obelix-out] type=peer auth=md5 host=voip.obelix.dnv.org username=asterix-in secret= context=international trunk=yes ### iax.conf from dogmatix server ; IAX Trunks [asterix-in] type=user auth=md5 host=voip.asterix.dnv.org secret= context=international trunk=yes [asterix-out] type=peer auth=md5 host=voip.asterix.dnv.org username=dogmatix-in secret= context=international trunk=yes The iax.conf from the obelix server would be similar. Hope this gives the idea OK - let me know if you need any more information. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 11-Mar-06, at 8:04 AM, Jean-Louis curty wrote: Hi, I successfully connected 2 servers via IAX but I'm pulling my hair to connect 2 extra servers , Anyone connected 3 or 4 servers together ? is it possible ? I d like to share the dialplan so _2 goes to server A _3 goes to serverB _4x goes to server C etc from the 4 servers any example of which one is peer, which one is user or friend would help me :-) thanks jl ___ --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] chan_zap ast_pickup_call issue redux
I'm running latest asterisk and zaptel, I have loaded wctdm and lsmod shows that it is in the kernel. I have configured the FXS and FXO ports on my TDM400P, and ztcfg shows both as configured with no errors. When I start asterisk I get the following error: Mar 13 14:07:41 WARNING[10958]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_cal. A search of the web and this mailing list shows issues related to the module not being loaded or zaptel not having been compiled before asterisk. I recompiled asterisk to ensure that it was linked against zaptel and manually deleted the previously installed version of chan_zap.so before doing make install. After following this resolution the issue persists with the same error as before. Any help in getting zaptel working would be greatly appreciated. ___ --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] priorityjumping=no
I've been trying to use a set-up whereby I have several TA's connected to an Asterisk server (1.2.4) and they act like they're in a hunt-group i.e. try the first, if busy jump to the next etc. in my extensions.conf I had something like [inbound-trunk] exten = 441234123456,1,Dial(SIP/s1a,20,r) exten = 441234123456,102,Dial(SIP/s2a,20,r) exten = 441234123456,203,Dial(SIP/s1b,20,r) exten = 441234123456,304,Dial(SIP/s2a,20,r) i.e. try the first, if busy try the next etc. It seemed to consistently fail. in [globals] priorityjumping=no was set, which came from the samples (i.e. make samples when installing Asterisk). I changed that to yes (i.e. priorityjumping=yes) and it started to work. If that was the problem (which it seems to be), is that the wrong default? Or am I missing something here completely? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Regexten Regcontext, working now
Just figured it out, I think. I put regcontext=mycontext into the [general] section in sip.conf instead of the the [user] section and when the sip user registered, the NoOp extension priority 1 came right up in the dial plan. All is well again, so far. Clarity of sight becomes infinitely greater with head removed from rectum. Hi All, I've been trying to get regexten and regcontext going for some sip peers but following the examples on the wiki is not working, as far as I can tell, nothing is happening. the phone registers, sip show peers is ok, but the NoOp priority 1 extension never gets created or added to the dialplan. Has anyone got this working? Thanks. JR JR Richardson Engineering for the Masses JR Richardson Engineering for the Masses ___ --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] Failed installing zaptel
Chuck Bunn wrote: Hi, I am having the same exact problem. I am assuming that it was a problem with a kernel update I did. I am in the process of rolling back to an older kernel... I will let you let know if this works. There is also a patch for zaptel but I believe this is for going from 1.3 to 1.4? If memory serves correctly, I think someone submitted a change to a makefile to handle an issue with recent kernels. Thought it was related to fc4 (or something like that), but might be what you're looking for. Think the change was submitted this weekend. ___ --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] Outgoing calls via Sipgate
Hello all, With some help from people in #asterisk on freenode, I've managed to get incoming SIP calls working. Outgoing calls however are however a different matter. My whole working (incoming calls only) SIPgate configuration can be found here. [1] When I uncommon what's in there, nothing works. There doesn't appear to be any useful error being logged , even when debug is enabled for console and file logs. If anyone could take a look and show me what needs adding in order for outgoing calls to work, that would be superb! My long term goal is to get asterisk running at home, and then persuade the boss to ditch the Avaya setup we have at the office. But since I'd likely be the one implementing it, I want to try and get something working before I commit myself :) Thanks!, Dave. [1] http://files.davehope.co.uk/home.tar ___ --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] Looking for docs on adjusting txgain/rxgain
Are these the droids you're looking for?: http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.ht ml I have corrected/edited the entry in the wiki. Also, is Kris Boutilier still around? Can anyone verify if this information has signifigantly changed in the last 18 months? Thanks, Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: Sunday, March 12, 2006 1:09 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Looking for docs on adjusting txgain/rxgain I am looking for docs on how to diagnose and adjust the rx/tx gain in zapata.conf. The wiki has a link to this article but it no longer exists on the server. http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.ht ml ___ --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] Cisco 7960 8.2 callerID lists proxy?
We rolled back to 7.4 cause of that too. 7.5 has a strange bug where if the server loses connection, the phone's just don't try re-registering. Aaron Tim Connolly wrote: Just curious, why not 7.5 ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Monday, March 13, 2006 2:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @proxy-addr... So the callerID on the phone looks like: [EMAIL PROTECTED] which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? Yeah, that bothered me so I rolled back to SIP 7.4. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] stop monitor on transfer
Ae you doing attended transfers or blind? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Monday, March 13, 2006 10:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] stop monitor on transfer Ive tried that via agis and that doesn't seem to work because the stopmonitor is applied to the call between the receptionist and the boss not the original call between caller and reception which is later transferred to the boss. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of Alexander |Lopez |Sent: Monday, March 13, 2006 9:25 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] stop monitor on transfer | |Setup a 'non-recording' extension for the oss and transfer the call to |that one. | |Ie: | |7123,1,StopMonitor |7123,2,Goto(123,1) | | | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] On Behalf Of Adrian | Carter | Sent: Monday, March 13, 2006 1:35 AM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: Re: [Asterisk-Users] stop monitor on transfer | | I'd teach the boss to appreciate recorded calls and just ensure they | are secure. | | I know mine actually loves that his calls are recorded - not many | people counter-claim or argue about conversations once you can trot | out them actually making the statement they claim they never did... | *shrug* | | horses for courses I guess - but other than the obvious (make em | appreciate and embrace rather than shun and dismiss) im not |sure what | you could do - Maybe just running stopmonitor again will stop the | first recording ? try just calling it twice on those calls | | Anton Krall wrote: | Guys. | | This idea has been banging my headfor days now and I feel | the need to | share with you. | | Imagine this scenario: all calls come in thru a | receptionist, asterisk | records all incoming calls, the receptionist's work is to | transfer the | calls to internal people but some of them are bosses and | you know how | bosses are, they don't want their calls to be recorded, so, I have | been trying to figure a way on how to stop monitoring / | recoring calls | once they are transferred to a bosses extension while othe | transferd | to other people stay on record mode. | | Anybody has done this or know of a way? | | I tried with stopmonitor but stopmonitor will stop | recording the call | between the receptionist and the boss but once the call is | transferred | and since the initial call come thru the recepcionist, the | call stays on record. | | What do you think guys? | | ___ | --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 | | | | -- | Adrian Carter | Technical Manager | Leading Edge Internet | | Web http://www.lei.net.au http://support.lei.net.au | Direct+61 2 6163 6162 Support 1 300 662 415 | E-mail[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 | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT call recording (was stop monitor on transfer)
On Mar 13, 2006, at 12:00 PM, Bob McDowell wrote: It depends http://www.callcorder.com/phone-recording-law-america.htm Thanks for the info! 12 states require, under most circumstances, the consent of all parties to a conversation. Those jurisdictions are California, Connecticut, Florida, Illinois, Maryland, Massachusetts, Michigan, Montana, Nevada, New Hampshire, Pennsylvania and Washington. I live in washington ;~) Marty ___ --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 Jitter Buffer for 1.2.5
Hi, I really want to start using 1.2.5, but I also really need to have the jitter buffer. Can anyone offer a suggestion of how to go? I've looked at the SIP_JB patch and there seems to be no indication of a patch for the 1.2.5 release. ___ --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
