Re: [asterisk-users] OT: USB T1/E1 Interface?
Michael Collins wrote: Just curious: has anyone seen or heard about a USB-based T1/E1 interface device? I’ve seen some serious T1/E1 testing equipment that is USB-based, but I was wondering if there was something more generic, like a Zaptel-ish T1/E1 that used USB instead of PCI/PCIx. Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Thanks! -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Met vriendelijke groeten, Remco Post SARA - Reken- en Netwerkdiensten http://www.sara.nl High Performance Computing Tel. +31 20 592 3000Fax. +31 20 668 3167 PGP Key fingerprint = 6367 DFE9 5CBC 0737 7D16 B3F6 048A 02BF DC93 94EC I really didn't foresee the Internet. But then, neither did the computer industry. Not that that tells us very much of course - the computer industry didn't even foresee that the century was going to end. -- Douglas Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Poor man's High Availability solution
On Thu, May 03, 2007 at 12:47:46AM +0200, Laurent Caron wrote: On Sun, Apr 29, 2007 at 09:06:53PM +0200, Clayton Milos wrote: Since a PRI is a physical connection as well as a logical one, if you can get the server to shut down when it has a problem you could put a 4-pole relay to change the PRI over to the other box. I think the ISDNGuard is more or less like a relay. Here is how I plan to set it up. Hook the Pri to the Net port of the ISDNGuard, hook the first port of each asterisk server's pri card to the 1st ast CPE and isdn CPE port, hook the second port of the pri card to the 2nd ast CPE and isdn CPE port, and finally plug the second Net port to the legacy PABX. Do you think this is the correct way of plugging two asterisk servers in front of a legacy pabx ? The attached file shows how it was supposed to be. attachment: ast.png___ --Bandwidth 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] rtpmap encoding parameters the 'unknown codec' problem?
We seem to have a problem with Asterisk 1.4 when a client sends through their SDP information but includes encoding parameters on the end of their SDP information. For example some phones send: a=rtpmap:18 G729/8000/1 instead of the usual: a=rtpmap:18 G729/8000 in the SDP... It seems that when the encoding parameter '/1' is included at the end of the rtpmap statement, Asterisk doesn't recognise the codec internally and then has trouble transcoding giving errors such as 'Unable to find a codec translation path from unknown to unknown'. 'sip show channels' also shows the 'Form' as 'unkn' during a call. This behaviour only appears to happen though when the encoding parameter is included. According to RFC2327: The general form of an rtpmap attribute is: a=rtpmap:payload type encoding name/clock rate[/encoding parameters] For audio streams, encoding parameters may specify the number of audio channels. This parameter may be omitted if the number of channels is one provided no additional parameters are needed. For video streams, no encoding parameters are currently specified. So, the encoding parameter part looks like an optional but perfectly valid part of the rtpmap SDP definition. Interestingly calls often seem to work fine out to the PSTN etc. but Asterisk has problems transcoding between 2 local clients. Has anybody seen this behaviour in Asterisk 1.4? Is this a bug or a feature that I haven't setup in Asterisk I am yet to discover? Cheers, Ray ___ --Bandwidth 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] Call Limit with multiple SIP extensions
Hello,
I´m trying to setup the following function fpr a customer and at the
moment I´m pretty stuck...
I have an Asterisk (realtime) system with about 50 SIP Phones. 4 of these
extensions have two SIP accounts behind them, so I call two SIP Accounts
via Dial(SIP/10SIP/11).
So for example I have the extension 20 which rings the SIP Accounts 20 and
*20 (callerid is setup to 20)
exten = 20,1,Dial(SIP/20SIP/*20)
This works so far. Now I need something like a call-limit function for
this special extension.
When one of the phones (20 or *20) is in use, the complete extension 20
should signal a busy to the next caller.
OK, i found the group function in asterisk and I managed to setup the
first half of the solution.
I use a macro
exten = 20,1,Macro(groupdialout,SIP/20SIP/*20,20)
[macro-groupdialout]
exten = s,1,Set(GROUP=${ARG2})
exten = s,n,Checkgroup(1)
exten = s,n,NoOP(Active Calls: ${GROUP_COUNT()})
exten = s,n,NoOp(${CHECKGROUPSTATUS})
exten = s,n,GotoIf($[${CHECKGROUPSTATUS} = OVERMAX]?100:6)
exten = s,n,Dial(${ARG1})
exten = s,100,Hangup(17)
to dial my extensions. This works on the incoming direction. But when I
dial an extension from one of my two phones the other phone is still
ringing. I also want a busy signal in this case.
I tried to increase the groupcounter on outgoing calls via:
exten = 20,1,Set(GROUP=${CALLERIDNUM})
and (to PSTN)
exten = _0X.,1,Set(GROUP=${CALLERIDNUM})
exten = _0X.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
But I think the problem is that group only affects the channel in use - am
I right?
Maybe someone can give me a hint on how to get this working? Or has
somebody did something like that before?
Thanks Michael
on the Asterisk CLI it looks like this so far:
-- Executing Set(SIP/20-08491430, GROUP=20) in new stack
-- Executing Macro(SIP/20-08491430, groupdialout|SIP/4450|4450) in
new stack
-- Executing Set(SIP/20-08491430, GROUP=4450) in new stack
-- Executing CheckGroup(SIP/20-08491430, 1) in new stack
-- Executing NoOp(SIP/20-08491430, Active Calls: 1) in new stack
-- Executing NoOp(SIP/20-08491430, OK) in new stack
-- Executing GotoIf(SIP/20-08491430, 0?100:6) in new stack
-- Goto (macro-groupdialout,s,6)
-- Executing Dial(SIP/20-08491430, SIP/4450) in new stack
-- Called 4450
-- SIP/4450-084ab3d0 is ringing
Extension Changed 4450 new state Ringing for Notify User 10
-- SIP/4450-084ab3d0 is ringing
-- SIP/4450-084ab3d0 is ringing
-- SIP/4450-084ab3d0 is ringing
-- SIP/4450-084ab3d0 is ringing
-- SIP/4450-084ab3d0 answered SIP/20-08491430
-- Attempting native bridge of SIP/20-08491430 and SIP/4450-084ab3d0
Extension Changed 4450 new state InUse for Notify User 10
-- Executing Set(SIP/10-0849d100, GROUP=20) in new stack
-- Executing Macro(SIP/10-0849d100, groupdialout|SIP/20SIP/30|20)
in new stack
-- Executing Set(SIP/10-0849d100, GROUP=20) in new stack
-- Executing CheckGroup(SIP/10-0849d100, 1) in new stack
-- Executing NoOp(SIP/10-0849d100, Active Calls: 1) in new stack
-- Executing NoOp(SIP/10-0849d100, OK) in new stack
-- Executing GotoIf(SIP/10-0849d100, 0?100:6) in new stack
-- Goto (macro-groupdialout,s,6)
-- Executing Dial(SIP/10-0849d100, SIP/20SIP/30) in new stack
-- Called 20
-- Called 30
-- SIP/30-084bfce0 is ringing
-- SIP/20-084b7d30 is ringing
So in this case it dows not work.
On incoming calls it seems to work...
-- Executing Set(SIP/10-08491430, GROUP=20) in new stack
-- Executing Macro(SIP/10-08491430, groupdialout|SIP/20SIP/30|20)
in new stack
-- Executing Set(SIP/10-08491430, GROUP=20) in new stack
-- Executing CheckGroup(SIP/10-08491430, 1) in new stack
-- Executing NoOp(SIP/10-08491430, Active Calls: 1) in new stack
-- Executing NoOp(SIP/10-08491430, OK) in new stack
-- Executing GotoIf(SIP/10-08491430, 0?100:6) in new stack
-- Goto (macro-groupdialout,s,6)
-- Executing Dial(SIP/10-08491430, SIP/20SIP/30) in new stack
-- Called 20
-- Called 30
-- SIP/30-0849a500 is ringing
-- SIP/20-084a7d68 is ringing
Extension Changed 20 new state Ringing for Notify User 10
-- SIP/20-084a7d68 is ringing
-- SIP/20-084a7d68 is ringing
-- SIP/30-0849a500 answered SIP/10-08491430
-- Attempting native bridge of SIP/10-08491430 and SIP/30-0849a500
Extension Changed 20 new state Idle for Notify User 10
Extension Changed 4450 new state InUse for Notify User 10
-- Executing Set(SIP/4450-084baaa0, GROUP=4450) in new stack
-- Executing Macro(SIP/4450-084baaa0,
groupdialout|SIP/20SIP/30|20) in new stack
-- Executing Set(SIP/4450-084baaa0, GROUP=20) in new stack
-- Executing CheckGroup(SIP/4450-084baaa0, 1) in new stack
-- Executing NoOp(SIP/4450-084baaa0, Active Calls: 2) in new stack
-- Executing NoOp(SIP/4450-084baaa0, OVERMAX) in new stack
-- Executing GotoIf(SIP/4450-084baaa0, 1?100:6) in new stack
-- Goto (macro-groupdialout,s,100)
Re: [asterisk-users] SIP Proxy
Ronaldo wrote: Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router http://www.voip-info.org/wiki/view/SIP+Express+Router: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org http://www.voip-info.org/wiki/view/Vovida.org * sipX http://www.voip-info.org/wiki/view/sipX from Sipfoundry http://www.voip-info.org/wiki/view/SIPfoundry is a native SIP proxy but also a complete SIP PBX * OpenSER http://www.voip-info.org/wiki/view/OpenSER - scalable and robust SIP server with TLS support Can anyone suggest me something about these SIP Proxy? SER and OpenSER are related. SER is more geared towards stability, OpenSER towards features. I guess that If SER does what you need, go with SER, otherwise look at OpenSER. I think that for instance freeworddialup uses SER. I'd think it is a very viable solution. The other two I don't knnow. p.s) Is Asterisk a SIP Proxy? No, it is not, * is a back2back user agent. Regards ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth 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] Reinvite after DTMF?
On 5/2/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Wilson Pickett [EMAIL PROTECTED] Date: Wed, 2 May 2007 15:30:21 +0200 Is there a way to do the following scenario? 1) my asterisk box receives an incoming call from a toll free number provider such as nufone, voicepulse, etc. 2) It then dials a number via SIP and outputs a DTMF sequence. At this point, I assume, the destination SIP has not been invited? The purpose of the DTMF is either determine which SIP destination to invite or to perform some other dial plan functions. ok, that part we do every day. 3) After DTMF though, is it possible to get the two SIP channels (original SIP caller plus SIP called) hooked together and have my pbx no longer in the call at all? tia If the above is true, then there shouldn't be a problem if all other conditions for reinvite are satisfied, because Asterisk will only execute Dial at this point, and that Dial could follow with reinvite. (I assume that the original SIP caller is in fact the toll free provider.) So what is in the dialplan once the DTMF is sent? The two channels are already bridged, how can asterisk then bow out? I don't see a way, but I thought I'd ask if someone else did? ___ --Bandwidth 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] allowing call every 15mins
Am Mittwoch, den 02.05.2007, 20:04 +0100 schrieb Goke Aruna: Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this. Are you trying to hint for future content of the Dilbert cartoon strip? What exactly is the point in allowing calls only during those four minutes? You would probably better set up a Queue system where callers can wait for anyone to answer the phone... or at least listen to some music NOT. By the way, once the customers learn about such an unusual setup, once problems start, the system will get a workload spike four times per hour - this is probably not something you want to achieve. (Have trouble with phones = get more trouble soon) As others write, this implementation is technically possible, and not even too complicated, but for the reason of it, I draw a blank. Would you please explain the idea behind it, for our all viewing pleasure? Regards, Anselm ___ --Bandwidth 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] [Announce] Web-MeetMe V3.0.1 released
Hello Dan, I finally got some time to test the SVN branches and here are my comments: One thing that does not work for sure - I had some problems to terminate the running conference from within the web page - I just clicked the button and nothing happened. This is likely a manager.conf security issue, but it could be a problem in the php code. I just tested branches/3.0 and trunk against 1.4.1 and it worked as expected. If you set core verbose to 10 and click on 'End Now' the console should display a message like this: app_meetme.c:941 meetme_cmd: Cmdline: 41251|k|1 At this point this would be a topic better suited for the support forums on SF. I browsed the php sources trying to understand and from what I see, the End Now button does nothing else than kicking all attendees - not exactly what I would expect. I would expect this action to terminate the conference immediately so, it could be seen in the Past conferences list. Also, some javascript popup dialog confirming this action would be nice. The same is valid for the Extend button - it works, but from the user prospective, nothing happens - I would expect some dialog box like The conference # has been extended by 10 minutes. This is the only missing piece, I would say - thanks :-) Ondrej The information contained in this e-mail and in any attachments is confidential and is designated solely for the attention of the intended recipient(s). If you are not an intended recipient, you must not use, disclose, copy, distribute or retain this e-mail or any part thereof. If you have received this e-mail in error, please notify the sender by return e-mail and delete all copies of this e-mail from your computer system(s). Please direct any additional queries to: [EMAIL PROTECTED] Thank 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] 0 duration but non-zero billsec in mysql cdr
I was just going through my call records ( stored in mysql database by cdr_MYSQL module ) and saw a record having duration = 0 and billsec of more than 50 seconds . I did a query on cdr where duration billsec and saw that there were infact some 250 records with duration less than billsecond ( table had around 4,00,000 records) . Did anyone came across this ? I also checked csv files and they had same record with duration 0 and higher bill seconds . Happen with both asterisk 1.2.17 as well as 1.2.18 All sip to iax/sip calls . Destination numbers were valid. Dialplan maintained with freepbx . ___ --Bandwidth 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] 0 duration but non-zero billsec in mysql cdr
That's a feature to generate more revenue. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Thursday, May 03, 2007 4:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 0 duration but non-zero billsec in mysql cdr I was just going through my call records ( stored in mysql database by cdr_MYSQL module ) and saw a record having duration = 0 and billsec of more than 50 seconds . I did a query on cdr where duration billsec and saw that there were infact some 250 records with duration less than billsecond ( table had around 4,00,000 records) . Did anyone came across this ? I also checked csv files and they had same record with duration 0 and higher bill seconds . Happen with both asterisk 1.2.17 as well as 1.2.18 All sip to iax/sip calls . Destination numbers were valid. Dialplan maintained with freepbx . ___ --Bandwidth 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] 0 duration but non-zero billsec in mysql cdr
Someone in -biz list pointed out that this could be a freepbx problem so i think i will go check there . @ Salvatore Giudice: how can i intentionally do it ? Damn i need a app that can make sure customer phone doesnt hangup for the time i specify .. even if customer breaks his phone . lol ___ --Bandwidth 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] UK - 2 port ISDN2e cards ...
Anyone have a quick recomendation for a 2-port (ie. 4 channels) BRI (ISDN2e) card in the UK? My usual source has 1 port and 4 port, but no 2-port cards... Thanks! Gordon ___ --Bandwidth 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] 0 duration but non-zero billsec in mysql cdr
Roflol. How about a script that makes calls for people after 15 min of inactivity... Streamline the whole process. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Thursday, May 03, 2007 4:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 0 duration but non-zero billsec in mysql cdr Someone in -biz list pointed out that this could be a freepbx problem so i think i will go check there . @ Salvatore Giudice: how can i intentionally do it ? Damn i need a app that can make sure customer phone doesnt hangup for the time i specify .. even if customer breaks his phone . lol ___ --Bandwidth 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] sipura spa9x1 - missed calls not wanted
Is there a way of cancelling the missed call entry on a Sipura 921/941 phone? E.g. when a call is signalled to three phones, one picks up - it's a nuisance having the other two list the call as missed. Is this something I can configure in *, or is it likely to be sipura-specific? /Per Jessen, Zürich ___ --Bandwidth 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] IP Phone Provisioning Tool by voip.com.sg - xml generation
San Singhania wrote: If you are interested in it you can download a copy at http://www.voip.com.sg/voip_ip_phone_provisioning_tool.html I got an HTTP 404 on the above. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth 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] Large dial plans and variables
You're so right! I thought about having just a catchall _. extension in the dialplan and doing everything else in a real language via AGI - PHP, Perl, ... whichever you like. It would make the programming part much easier as the scope of variables is just as you expect it to be. Well, they're called macro's for a reason You guys are proposing adding functions or procedures. My first step in any macro would be to copy incoming variables, be it arguments or even asterisk defined stuff to local variables. But that is just me and my coding convention. -- Andreas Sikkema ___ --Bandwidth 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] UK - 2 port ISDN2e cards ...
Gordon Henderson wrote: Anyone have a quick recomendation for a 2-port (ie. 4 channels) BRI (ISDN2e) card in the UK? My usual source has 1 port and 4 port, but no 2-port cards... http://www.beronet.com lists a 2-port BRI card, but I don't know about availability in the UK; maybe they don't mind shipping to the UK. (I have not tried any of these cards.) (aren't you guys getting rid of ISDN anyway? :-) /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth 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] Get Asterisk to redirect a SIP INVITE
I want to get Asterisk to redirect an incoming SIP INVITE to another SIP URI. I was looking at the Transfer application but it seems to be broken (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483). Is there an alternative way to do this on Asterisk 1.2.18? Regards Cameron ___ --Bandwidth 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] Double DTMF digits sent on IAX native bridge
This is very interesting. I am now getting this double-digit behaviour occasionally, and only on IAX channels (so far). Did anyone come up with a solution or a way to improve matters? The scenario where I get this is: PSTN - Provider - IAX - Gateway - IAX - Customer So I will go and do some debugging to see where the extra packets are generated. Perhaps a simple change to prevent DTMF audio-tone detection on a channel that supports RFC or out-of-band DTMF transmission? Or perhaps some sort of DTMF debounce mechanism? Thanks, Steve On 3/6/07, Remi Quezada [EMAIL PROTECTED] wrote: Any ideas as to how I can fix this issue? Thanks Remi Remi Quezada wrote: Ok that makes sense, but I'm still getting double digits. It seems to me that the DTMF digit is getting detected too late. When the digit is pressed it seems like asterisk is passing the DTMF digit for a fraction of a second through the audio path and then sends the digit for however long your toneduration is set for. I can hear this happening when I dial the digits myself, I hear some kind sound being cut off for a fraction of a second and then hear the DTMF tone pass. So I guess this is why sometimes some answer machines are detecting double digits. Russell Bryant wrote: Remi Quezada wrote: I have two asterisk servers one is connected to the PSTN and the other one is connected to SIP users. The two servers connect with each other using IAX. When I have an incoming call from PSTN to the asterisk servers and have a forward to go back out to the PSTN the two IAX channel bridge together. Now every time I dial a DTMF digit, the asterisk is sending two DTMF digits. I enable debugging for iax and I do see it sending the DTMF digits two times. Here is what I see: The IAX debug that you show below only shows one of each digit. For each one, it shows Receiving the digit from one leg of the call, and then transmitting it out the other. I have spaced out your debug to separate each digit. Each one shows ... - digit - ACK - - digit --- ACK -- which is exactly what is supposed to happen. ___ --Bandwidth 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] sipura spa9x1 - missed calls not wanted
On Thu, 2007-05-03 at 11:35 +0200, Per Jessen wrote: Is there a way of cancelling the missed call entry on a Sipura 921/941 phone? E.g. when a call is signalled to three phones, one picks up - it's a nuisance having the other two list the call as missed. Is this something I can configure in *, or is it likely to be sipura-specific? No. The phone in question didn't answer the call so it's a missed call as far as it's concerned. -- Dave Cotton [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] IP Phone Provisioning Tool by voip.com.sg - xml generation
On Thu, 2007-05-03 at 11:38 +0200, Per Jessen wrote: San Singhania wrote: If you are interested in it you can download a copy at http://www.voip.com.sg/voip_ip_phone_provisioning_tool.html I got an HTTP 404 on the above. By going to the main pageand trying asterisk stuff I found this. http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html -- Dave Cotton [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] ADSL routers with integrated SIP QoS for otherdevices
In my experience I'm using a comtrend CT-536+ It's a broadcom 96348GW model 266Mhz Mips r4K compliant CPU 16MB RAM and 16 MB flash, adsl2+, 4 port ethernet switch, usb 1.1 and .g type wireless. I'm using an Asus WL600g based firmware because de uclibc and toolchain version of cross compiler, but there are others with diferent capabilities like Ipsec etc.. I know a t least of Telsey one. Mine is a 2.6.8.1 Kernel version with time infraestructure for mips backported from 2.6.16 kernel (High resolution timers), I have to check if it's worth... It has Iproute2 and iptables and ebtables, dscp capable and diff-serv qos. Zaptel 1.4.1 with only ztdummy zttest and libtonezone, and a mini version of asterisk 1.4.2 but I'm working on a sqlite2 and realtime capable one... No h.323, not moh although the usb can be used for a memory stick, ah... It's used for callshops with yuxins sip/iax phones, no TRANSCODING just g.729 end to end, we use this to bridge sip-iax2/IAX2--Internet--IAX2/SIP---Lucent Compact Switch--Intl PSTN Carriers because of trunking capabilities El Miércoles, 2 de Mayo de 2007 14:28, Mike Dent escribió: On 5/2/07, Tim Koehler [EMAIL PROTECTED] wrote: Hi, I can agree for smaller installation/home offices the Linksys WRT series is pretty good (I'm using this at home). I'm using the dd-wrt Firmware (www.dd-wrt.com ) which is also available for plenty other routers. With QoS values set right I always have clear audio even under rough conditions (sending big mails, receiving a joost stream, big download + VoIP call). DD-WRT even is available with a SIP Proxy and a VPN Server/Client. For bigger installations I can recommend the - Borderware SIPAssure, - InGate solutions - Intertex routers. All fully SIP-aware and with more ore less additional featureset (DoS filter, SRTP+TLS, SPIT filter, etc.). Cheers Tim --- snom technology AG Tim Koehler Partner Manager [EMAIL PROTECTED] Tim, great to see your participation in the Asterisk list and nice to see you using the dd-wrt firmware. Might I ask which model WRT you are using and if you use a seperate ADSL modem which one? Thanks Mkke ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Francisco J. Pérez Botella ___ --Bandwidth 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] IP Phone Provisioning Tool by voip.com.sg - xmlgeneration
If you are interested in it you can download a copy at http://www.voip.com.sg/voip_ip_phone_provisioning_tool.html I got an HTTP 404 on the above. A very quick look on the site and i found the URL from the Menus.. http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html Best Regards Gavin Spurgeon Assistant Systems Administrator Leigh City Technology College [EMAIL PROTECTED] http://www.leighctc.kent.sch.uk Tel: 01322 620501 Fax: 01322 620599 IS HelpDesk : Ext 541 -- This message has been scanned for viruses and dangerous content by the Systems @ the LeighCTC, 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] sipura spa9x1 - missed calls not wanted
Dave Cotton wrote: On Thu, 2007-05-03 at 11:35 +0200, Per Jessen wrote: Is there a way of cancelling the missed call entry on a Sipura 921/941 phone? E.g. when a call is signalled to three phones, one picks up - it's a nuisance having the other two list the call as missed. Is this something I can configure in *, or is it likely to be sipura-specific? No. The phone in question didn't answer the call so it's a missed call as far as it's concerned. Yeah, that's what I was afraid of. Is there really no way around this? It seems like it is something many people could be running into. /Per -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth 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] My Polycom IP 501 is formatted its file system itself
Hi, Thank you for your response. My phone is giving boot menu and giving a chance to load firmware image. How can do this? Can you please send me those boot files and configuration procedure please? Look forward to your response. Thank you. Regards, Chandra. Noah Miller [EMAIL PROTECTED] wrote: Hi Chandra - We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. - 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 - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --Bandwidth 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] My Polycom IP 501 is formatted its file systemitself
Hi, Thank you for your response. My phone is giving boot menu and giving a chance to load firmware image. How can do this? Can you please send me those boot files and configuration procedure please? Look forward to your response. Thank you. Regards, Chandra. Steve Totaro [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Wednesday, April 25, 2007 9:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] My Polycom IP 501 is formatted its file systemitself Hi Chandra - We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. - Noah If it just keeps rebooting with an error about not being able to load the application or something like that, I had one do that. I found a fix via google. If this is the issue, post back and I will see if I can find the link. It is an easy fix. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --Bandwidth 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] My Polycom IP 501 is formatted its file system itself
Hi, Thank you for your response. My phone is giving boot menu and giving a chance to load firmware image. How can do this? Can you please send me those boot files and configuration procedure please? Look forward to your response. Thank you. Regards, Chandra. »Steven Ringwald« [EMAIL PROTECTED] wrote: Noah Miller wrote: Hi Chandra - We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine except one phone. When I tried to connect my phone with my network, It automatically formatted its file system. Now, It is not booting. What I have to do now? Can you please tell me the solution. What is it doing? Do you get a boot menu at all? Is it totally dead (won't power up)? Are you using a boot server (TFTP, FTP, HTTP, HTTPS)? If it's totally dead, you'll want to speak with your Polycom reseller. They should replace it for you. If the phone boots, and you can get into the boot menu, it may be that there is a configuration option in the boot menu that is preventing the phone from talking to your boot server. If you can get to the boot menu (where it offers to let you configure a server to boot off of), you can recover with a firmware image. You usually can get these only from resellers (because Polycom doesn't want to deal with customer support on an individual basis). Let me know if you can get this far; I might be able to help. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --Bandwidth 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] CDR(accountcode) empty in * 1.4.4 (for local chan)
Hi all,
i just updated to asterisk 1.4.4 from 1.4.2. i was doing this to forward an
unanswered call in 1.4.2
exten= 1,1,Dial(SIP/123,,Ttg)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10)
exten= 1,3,Hangup
exten= 1,10,Dial(Local/2,,Ttg)
exten= 1,11,Hangup
exten= 2,1,Dial(SIP/234,,Ttg)
exten= 2,2,Hangup
All the CDR variables for the first channel (SIP/123) are fine. but when
local channel initiates, it does not copy the CDR(accountcode) variable from
the first channel (in asterisk 1.4.4), whereas it did in 1.4.2. so the
CDR(accountcode) variable for local channel is empty in 1.4.4. This is a big
problem for me as i have to charge the forwarded calls also and all calls
are charged based on account code. If accountcode is empty, i cant make a
decision how to charge the call.
Can anybody fix this for me or do i have to jump back to asterisk 1.4.2?
--
Regards
Rizwan Hisham
Software Engineer
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Re: [asterisk-users] VPN between Asterisk server and phone client
Salvatore Giudice wrote: Any network service could potentially harbor a buffer overflow, etc that could result in remote command execution. Provided someone find a similar bug and it's exploitable, they would theoretically be able to spawn a shell with the same rights as Asterisk. Generally, it's better to run services as nobody. I would be hesitant to allow management of VPN's from within Asterisk. Check out this link: http://mixter.void.ru/exploit.html It's a basic tutorial on writing shell code for buffer overflows. The basic idea is you find some condition where you can cause the application to seg fault and if you are lucky, it will allow you to write your shell code to memory, gain control of the stack pointer, and make your shell code run. These types of exploits have to be tailored to specific OS's and architectures. Shellcode that works on a BSD system will not work on Solaris or Redhat, etc... Generally you can reuse the delivery code by swapping out the shell code for whatever you are attacking. I'm not stating these currently exist in Asterisk, but theoretically it is likely and we just don't know about it yet. Prudence suggest that we don't help the hackers any more than we have to in case they find it first. I think it would be really difficult to lockdown VPN if Asterisk manages it's operation. Asterisk would have to have execution rights to the VPN binaries or an intermediate script at the very least. Just my 2 cents. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Wednesday, May 02, 2007 8:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VPN between Asterisk server and phone client On 5/2/07, Salvatore Giudice [EMAIL PROTECTED] wrote: If you run it on the fly, doesn't that mean that the Asterisk user will have permissions to configure VPN's? Nobody sees a problem with that? I thinking that if you knock over the Asterisk service and get shell execution rights as Asterisk, you could be able to start tunnels for things other than voice. It's like giving a hacker a great way to hide their activities from your IDS without having to bother to get root first to install an encrypted data pipe. That's true, the asterisk user needs to be able to invoke the start_vpn script or program. That does not mean that the asterisk user will have to have superuser rights to configure VPNs. You could make the start_vpn program setuid to a user that has those rights (and in that case, you probably don't want start_vpn to be a script). Also, openvpn typically starts predefined VPNs. To define a new one, someone would have to have access to the file system. When you say knock over the Asterisk servoce and get shell execution rights, how would that happen, exactly? I can think of DoS attacks and other stuff, but am wondering how knocking over Asterisk will give someone shell execution rights? As I said above, you would want to make the function to start a VPN connection as safe as possible. That would include NOT using scripts, and employing other verification methods. One approach if you are really concerned and fit certain requirements. Just deny all unsolicited traffic on the normal IP except on whatever port you are running OpenVPN on and accept traffic on the OpenVPN IP. Thanks, Steve Totaro www.asteriskhelpdesk.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Daemontools and holidays macro
You may want to consider renaming daemontools as it is also the name of a windows program that allows you to mount CD/DVD ISOs, so there could be some confusion. On 5/2/07, Steve Totaro [EMAIL PROTECTED] wrote: Vicente Aguilar wrote: Hi I've recently released the daemontools scripts I use to run both Asterisk and Flash Operator Panel, and a macro to tell whether today is a holiday or not and jump to different dialplan places accordingly. They are here: daemontools scripts: http://www.bisente.com/blog/2007/04/27/spanish-asterisk-y-daemontools-spanishenglish-asterisk-and-daemontools-english/?lan=english is-holiday macro: http://www.bisente.com/blog/2007/04/30/asterisk-holidays/?lan=english Hope you find them useful. Any feedback or improvements will be appreciated. :) Regards, Thanks, Email moved to my Useful Asterisk Stuff Folder Thanks, Steve Totaro www.asteriskhelpdesk.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: USB T1/E1 Interface?
Remco Post wrote: Michael Collins wrote: Just curious: has anyone seen or heard about a USB-based T1/E1 interface device? I’ve seen some serious T1/E1 testing equipment that is USB-based, but I was wondering if there was something more generic, like a Zaptel-ish T1/E1 that used USB instead of PCI/PCIx. Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Maybe he is looking at making his own very portable testing equipment? An idea I entertained myself. A T-Bird is pretty expensive. An OpenSource T1 tester would probably sell pretty well, or at least generate alot of interest. I have seen PCMCIA to PCI adapters but they were quite expensive at the time (~$500). Maybe a shuttle or similar matx case with an LCD would work well. Thanks, Steve Totaro www.asteriskhelpdesk.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] zttranscode crashes server
I was just looking to see if anyone else has seen this problem as well. When asterisk starts up it loads the zttranscode module. The problem exist when I use the init scripts to stop asterisk and then use the zaptel init script to unload modules. Since the zaptel init script didn't load the zttranscode module it will error out when trying to unload the modules. I built zaptel/asterisk/libpri (all latest releases as of May 3rd). I am also using the /etc/sysconfig/zaptel file to only specify the two modules I do need. wct4xxp and wctdm24xxp. I am using a TE212P and a TDM844B card. I shouldn't need the zttransode module since I don't have a codec translation card. right? To work around this I added zttranscode to RMODULES in the zaptel init script. If I don't need the zttranscode module. I may try and rebuild zaptel without it. -- *** Forrest Beck IAXTEL: 17002871718 [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] UK - 2 port ISDN2e cards ...
On Thu, 3 May 2007, Per Jessen wrote: Gordon Henderson wrote: Anyone have a quick recomendation for a 2-port (ie. 4 channels) BRI (ISDN2e) card in the UK? My usual source has 1 port and 4 port, but no 2-port cards... http://www.beronet.com lists a 2-port BRI card, but I don't know about availability in the UK; maybe they don't mind shipping to the UK. (I have not tried any of these cards.) Hi, I've found a place in Germany which stocks them. Thanks! ( http://www.asteriskcards.eu/ if anyon else is intersted) (aren't you guys getting rid of ISDN anyway? :-) H... Some people would like to think so, but it's going to be here for a long time yet! BT have/are dumping the consumer versions of ISDN2 - home highway which went a while back, but business highway is going soon if it's not already gone, which is a real shame as they had almost all the functionality a small business needed for less price than the full ISDN2e... Cheers, Gordon ___ --Bandwidth 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] zttranscode crashes server
On Thu, May 03, 2007 at 08:38:10AM -0400, Forrest Beck wrote: I was just looking to see if anyone else has seen this problem as well. When asterisk starts up it loads the zttranscode module. The problem exist when I use the init scripts to stop asterisk and then use the zaptel init script to unload modules. Since the zaptel init script didn't load the zttranscode module it will error out when trying to unload the modules. I built zaptel/asterisk/libpri (all latest releases as of May 3rd). I am also using the /etc/sysconfig/zaptel file to only specify the two modules I do need. wct4xxp and wctdm24xxp. Strangely enough, this issue exists in 1.4, but not in 1.2. Compare: http://svn.digium.com/svn/zaptel/branches/1.2/zaptel.init http://svn.digium.com/svn/zaptel/branches/1.4/zaptel.init Note the unload_module function in 1.2 (yicks: recursive functions in bourne shell) I am using a TE212P and a TDM844B card. I shouldn't need the zttransode module since I don't have a codec translation card. right? Right. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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] Daemontools and holidays macro
On Thu, May 03, 2007 at 07:04:59AM -0500, William Moore wrote: You may want to consider renaming daemontools as it is also the name of a windows program that allows you to mount CD/DVD ISOs, so there could be some confusion. And in a closer context, daemontools is also one of DJB's creations: http://cr.yp.to/daemontools.html -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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] Digital Phones
I've used these gateways and never experienced any of these problems. I could imagine me missing the popping noise but I do know that MWI did work just fine. Steve Totaro wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, May 02, 2007 1:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digital Phones Salvatore Giudice wrote: Nortel digital Meridian phones are like $400/each. At least that was the price of the phones at a hotel I did a job for recently. Still? (Is Nortel even making these phones anymore? I thought they spun off their telephone set division -- anybody heard of Aastra? ;) ) When you go to SIP, you may save on the capital costs for the phones, but other costs will increase. These are related to: 1.) increased support requirements for supporting VoIP 2.) additional licensing may be required from your vendor to support SIP or IP media 3.) increased costs associated with mitigating potential damages since your voice services are now subject to the same outages as your network 4.) increased training costs for staff to become proficient in VoIP 5.) increased costs associated with monitoring QoS 6.) increased costs associated with reconfiguring your network for VoIP and QoS - many times new switches may have to be purchased in addition to SBC's etc 7.) additional costs associated with rewiring physical space to accommodate additional Ethernet ports required for phones Yes, 7 times. Sometimes, if you already have digital - it may not be worth switching to SIP even if you save a ton on the handsets. Whenever we switch over a hotel to VoIP, we always run into these extra 'hidden' costs. If you want to do digital with asterisk, I think you'll need a T1/E1 multiplexer that supports digital phones. Is this anything like a channel bank, only for digital phones? Can you suggest any examples? -Stephen- Citel makes SIP to Digital gateways. I have had poor experience with them and doubt I would try it again without seeing many improvements listed in their firmware releases. Just to clarify, I had loud bursts of static when first picking up or originating a call, phones resetting for no reason, regular interval popping sounds. Citel fixed the popping sound by replacing one of the gateways but the other issues were never fixed. Also, the phones did not act the way I would have liked. When dialing, only a beep was heard, not actual DTMF and stutter dialtone MWI could not be disabled even though the phone I used had a MWI light and LCD display showed messages. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth 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] Answer A Ringing Queue By Dialing An Extension
Stick the lobby phones into a call group and put your other phones in that pickup group. Then you can hit *8 to pick up those calls (or, if you have speed dial/BLF/softkeys, program one of those as *8 for an immediately accessible button). In sip.conf for the lobby phones: callgroup=1 in sip.conf for the other phones: pickupgroup=1 Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth 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] Called party identification - where to take called name?
Hello, I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. Thanks, __Yehavi: ___ --Bandwidth 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] Virtual IP Adresses and SIP requests failing...
Hey All: Question; when using a virtual IP on an Asterisk server, I am having trouble getting sip user to register to the ViP. They are able to register with the true IP, just not the virtual. It seems Asterisk is rejecting the SIP invite, register, etc (like it's not destined for this server) I've added all the IP's to the domain listing in sip.conf and in the Asterisk console a sip show domains shows both the virtual and the physical IP. Am I missing something? I have Asterisk bound to 0.0.0.0 which should tell it to listen on all IP's, right?? Some Details: ## ifconfig eth1 - inet addr:69.67.250.38 eth1:0 - inet addr:69.67.250.36 (ViP) ## sip.conf [general] domain=69.67.250.36 domain=69.67.250.38 bindport=5060 port=5060 bindaddr=0.0.0.0 ## sip show domains Our local SIP domains: Context Set by 69.67.250.36 (default) [Configured] 69.67.250.38 (default) [Configured] ## tshark -i eth1 -R sip ## Call to .38 10.818719 66.218.1.47 - 69.67.250.38 SIP Request: OPTIONS sip: 69.67.250.38 10.818903 69.67.250.38 - 66.218.1.47 SIP Status: 200 OK 10.820676 192.168.0.102 - 69.67.250.38 SIP Request: OPTIONS sip: 69.67.250.38 10.821626 69.67.250.38 - 192.168.0.102 SIP Status: 200 OK 10.829019 66.218.1.47 - 69.67.250.38 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 10.830792 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 10.835473 66.218.1.47 - 69.67.250.38 SIP Request: ACK sip:[EMAIL PROTECTED] 10.841651 66.218.1.47 - 69.67.250.38 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 10.841880 69.67.250.38 - 66.218.1.47 SIP Status: 100 Trying 10.847744 69.67.250.38 - 69.67.248.83 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 10.847874 69.67.250.38 - 66.218.1.47 SIP/SDP Status: 183 Session Progress, with session description 10.848852 69.67.248.83 - 69.67.250.38 SIP Status: 100 Trying 16.724167 69.67.248.83 - 69.67.250.38 SIP/SDP Status: 183 Session Progress, with session description 16.725928 69.67.248.83 - 69.67.250.38 SIP/SDP Status: 200 OK, with session description 16.726053 69.67.250.38 - 69.67.248.83 SIP Request: ACK sip:[EMAIL PROTECTED]:5060 16.726373 69.67.250.38 - 66.218.1.47 SIP/SDP Status: 200 OK, with session description 16.731913 66.218.1.47 - 69.67.250.38 SIP Request: ACK sip:[EMAIL PROTECTED] 19.561514 69.67.248.83 - 69.67.250.38 SIP Request: BYE sip:[EMAIL PROTECTED] 19.561617 69.67.250.38 - 69.67.248.83 SIP Status: 200 OK 19.562158 69.67.250.38 - 66.218.1.47 SIP Request: BYE sip:[EMAIL PROTECTED]:5004;transport=udp 19.565798 66.218.1.47 - 69.67.250.38 SIP Status: 200 OK ## Call to .36 90.821676 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 90.821873 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 91.321664 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 91.822061 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 92.322452 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 92.821931 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 94.323765 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 94.452850 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 94.453240 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 94.822695 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 98.324204 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 98.453399 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 98.822235 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 102.325048 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 102.821775 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 106.325130 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 106.822293 69.67.250.38 - 66.218.1.47 SIP Status: 407 Proxy Authentication Required 110.326101 66.218.1.47 - 69.67.250.36 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 110.587025 66.218.1.47 - 69.67.250.36 SIP Request: CANCEL sip:[EMAIL PROTECTED] 110.587101 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 110.587133 69.67.250.38 - 66.218.1.47 SIP Status: 200 OK 111.087270 66.218.1.47 - 69.67.250.36 SIP Request: CANCEL sip:[EMAIL PROTECTED] 111.087332 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 111.087359 69.67.250.38 - 66.218.1.47 SIP Status: 200 OK 111.587718 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request Terminated 112.087661 69.67.250.38 - 66.218.1.47 SIP Status: 487 Request
[asterisk-users] Semi-OT: useful things to do with XML browsers in phones
Greetings list, It seems that more and more phones these days are coming with XML mini-browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. So I'm open to suggestions. What useful applications are you developing for these mini-browsers? What sort of things do your customers want to use on them? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth 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] Wildcard TE410P problem
Hello all, I'm using Wildcard TE410P card. Here is a zaptel.cfg loadzone=se defaultzone=se span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 span=2,0,0,ccs,hdb3,crc4 bchan=32-46 dchan=47 bchan=48-62 span=3,0,0,ccs,hdb3,crc4 bchan=63-77 dchan=78 bchan=79-93 span=4,0,0,ccs,hdb3,crc4 bchan=94-108 dchan=109 bchan=110-124 # ztcfg -vvv Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) Channel 32: Clear channel (Default) (Slaves: 32) Channel 33: Clear channel (Default) (Slaves: 33) Channel 34: Clear channel (Default) (Slaves: 34) Channel 35: Clear channel (Default) (Slaves: 35) Channel 36: Clear channel (Default) (Slaves: 36) Channel 37: Clear channel (Default) (Slaves: 37) Channel 38: Clear channel (Default) (Slaves: 38) Channel 39: Clear channel (Default) (Slaves: 39) Channel 40: Clear channel (Default) (Slaves: 40) Channel 41: Clear channel (Default) (Slaves: 41) Channel 42: Clear channel (Default) (Slaves: 42) Channel 43: Clear channel (Default) (Slaves: 43) Channel 44: Clear channel (Default) (Slaves: 44) Channel 45: Clear channel (Default) (Slaves: 45) Channel 46: Clear channel (Default) (Slaves: 46) Channel 47: D-channel (Default) (Slaves: 47) Channel 48: Clear channel (Default) (Slaves: 48) Channel 49: Clear channel (Default) (Slaves: 49) Channel 50: Clear channel (Default) (Slaves: 50) Channel 51: Clear channel (Default) (Slaves: 51) Channel 52: Clear channel (Default) (Slaves: 52) Channel 53: Clear channel (Default) (Slaves: 53) Channel 54: Clear channel (Default) (Slaves: 54) Channel 55: Clear channel (Default) (Slaves: 55) Channel 56: Clear channel (Default) (Slaves: 56) Channel 57: Clear channel (Default) (Slaves: 57) Channel 58: Clear channel (Default) (Slaves: 58) Channel 59: Clear channel (Default) (Slaves: 59) Channel 60: Clear channel (Default) (Slaves: 60) Channel 61: Clear channel (Default) (Slaves: 61) Channel 62: Clear channel (Default) (Slaves: 62) Channel 63: Clear channel (Default) (Slaves: 63) Channel 64: Clear channel (Default) (Slaves: 64) Channel 65: Clear channel (Default) (Slaves: 65) Channel 66: Clear channel (Default) (Slaves: 66) Channel 67: Clear channel (Default) (Slaves: 67) Channel 68: Clear channel (Default) (Slaves: 68) Channel 69: Clear channel (Default) (Slaves: 69) Channel 70: Clear channel (Default) (Slaves: 70) Channel 71: Clear channel (Default) (Slaves: 71) Channel 72: Clear channel (Default) (Slaves: 72) Channel 73: Clear channel (Default) (Slaves: 73) Channel 74: Clear channel (Default) (Slaves: 74) Channel 75: Clear channel (Default) (Slaves: 75) Channel 76: Clear channel (Default) (Slaves: 76) Channel 77: Clear channel (Default) (Slaves: 77) Channel 78: D-channel (Default) (Slaves: 78) Channel 79: Clear channel (Default) (Slaves: 79) Channel 80: Clear channel (Default) (Slaves: 80) Channel 81: Clear channel (Default) (Slaves: 81) Channel 82: Clear channel (Default) (Slaves: 82) Channel 83: Clear channel (Default) (Slaves: 83) Channel 84: Clear channel (Default) (Slaves: 84) Channel 85: Clear channel (Default) (Slaves: 85) Channel 86: Clear channel (Default) (Slaves: 86) Channel 87: Clear channel (Default) (Slaves: 87) Channel 88: Clear channel (Default) (Slaves: 88) Channel 89: Clear channel (Default) (Slaves: 89) Channel 90:
Re: [asterisk-users] Semi-OT: useful things to do with XML browsers in phones
On Thursday 03 May 2007 10:18 am, Chris Bagnall wrote: It seems that more and more phones these days are coming with XML mini-browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. What useful applications are you developing for these mini-browsers? What sort of things do your customers want to use on them? Queue stats, sales/donation volumes, weather/stock/news, door/gate alarms... -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] zttranscode crashes server
So is anyone not using the zaptel init script to load modules? Anyone using modules.conf? How an I load them at boot without using the init script? Do I just remove --ignore-install from modprobe? Thanks On 5/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, May 03, 2007 at 08:38:10AM -0400, Forrest Beck wrote: I was just looking to see if anyone else has seen this problem as well. When asterisk starts up it loads the zttranscode module. The problem exist when I use the init scripts to stop asterisk and then use the zaptel init script to unload modules. Since the zaptel init script didn't load the zttranscode module it will error out when trying to unload the modules. I built zaptel/asterisk/libpri (all latest releases as of May 3rd). I am also using the /etc/sysconfig/zaptel file to only specify the two modules I do need. wct4xxp and wctdm24xxp. Strangely enough, this issue exists in 1.4, but not in 1.2. Compare: http://svn.digium.com/svn/zaptel/branches/1.2/zaptel.init http://svn.digium.com/svn/zaptel/branches/1.4/zaptel.init Note the unload_module function in 1.2 (yicks: recursive functions in bourne shell) I am using a TE212P and a TDM844B card. I shouldn't need the zttransode module since I don't have a codec translation card. right? Right. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [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] Secondary redirect failed
hello all,i will make a call to asterisk server, that time the end user in ringing phase.After that i am trying to \redirect\ the call during ringing phase.This time the server shutdown...i want to answer the call during ringing phase.please help me if anyone knows.Regards,Pandi.P___ --Bandwidth 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] zttranscode crashes server
sorry, I meant modprobe.conf On 5/3/07, Forrest Beck [EMAIL PROTECTED] wrote: So is anyone not using the zaptel init script to load modules? Anyone using modules.conf? How an I load them at boot without using the init script? Do I just remove --ignore-install from modprobe? Thanks On 5/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, May 03, 2007 at 08:38:10AM -0400, Forrest Beck wrote: I was just looking to see if anyone else has seen this problem as well. When asterisk starts up it loads the zttranscode module. The problem exist when I use the init scripts to stop asterisk and then use the zaptel init script to unload modules. Since the zaptel init script didn't load the zttranscode module it will error out when trying to unload the modules. I built zaptel/asterisk/libpri (all latest releases as of May 3rd). I am also using the /etc/sysconfig/zaptel file to only specify the two modules I do need. wct4xxp and wctdm24xxp. Strangely enough, this issue exists in 1.4, but not in 1.2. Compare: http://svn.digium.com/svn/zaptel/branches/1.2/zaptel.init http://svn.digium.com/svn/zaptel/branches/1.4/zaptel.init Note the unload_module function in 1.2 (yicks: recursive functions in bourne shell) I am using a TE212P and a TDM844B card. I shouldn't need the zttransode module since I don't have a codec translation card. right? Right. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] -- *** Forrest Beck IAXTEL: 17002871718 [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] Daemontools and holidays macro
El jue, 03-05-2007 a las 07:04 -0500, William Moore escribió: You may want to consider renaming daemontools as it is also the name of a windows program that allows you to mount CD/DVD ISOs, so there could be some confusion. daemontools is not the name of my scripts, but the name of a program they use to ensure Asterisk is running 100% of the time: http://cr.yp.to/daemontools.html *This* daemontools spawn a process and monitor it, re-running it if it dies. Quite useful for high-availability services. -- Vicente Aguilar [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] Semi-OT: useful things to do with XML browsers inphones
Hi Chris, I'd also like to see more development in this area. It's continually disappointed me that more cross platform information delivery applications aren't being developed on asterisk. When I paid the first bounty to someone to write the weather text to speech routine it wasn't because someone asked for it - the idea came to me because I knew that Asterisk had FTP capabilities and that [EMAIL PROTECTED] had just introduced text to speech so I was trying to work out what could this speech functionality be used for. I knew that the NOAA had these flat text files readily available which is how come I paid someone $50 to write the now infamous weather routine. At the time I did this I thought it was the start of something new and wonderful and that a slew of Text to Speech routines would be written for Asterisk.. As far as I know I haven't seen a single one since. Flat text file information for stocks, indices, exchange rates, sports etc are readily availableI guess no one has seen a file they wish to access via audio for the moment (or they have and haven't posted the code to share) So my offer to you is this; If you can write a code and then gpl it to the Asterisk community to stream a ftp sourced (or some other transfer command driven by the display or keypad) txt file onto the handset browser page for at least 3 brands of handsets (Polycom must be one of them) then I'll start the bounty of $100 So basically write a code that I can download the NY weather txt file from NOAA and display the txt on my polycom 501 and I'll paypal you or anyone else $100 Some other cool files would be sunrise/sunset. I can basically set up my wifes handset on the trixbox server in our home to display this and no need for her to check the weather channel each morning. You might also want to check out Adhearsion.com as I understand Jay has been writing some code to make browser displays cross brand compatible. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Thursday, 3 May 2007 10:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Semi-OT: useful things to do with XML browsers inphones Greetings list, It seems that more and more phones these days are coming with XML mini- browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. So I'm open to suggestions. What useful applications are you developing for these mini-browsers? What sort of things do your customers want to use on them? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth 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] you have been kicked my this conference
How do I stop the you have been kicked by this conference message from speaking? I first had MeetMe(conf, l) and I get the kicked message. I tried Meetme(CONF, lq) and I still get he kicked message. and it still says it. Thanks, Jerry ___ --Bandwidth 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] Double DTMF digits sent on IAX native bridge
Replying to my own post... Even more interesting is that the issue seems to be caused by the Linksys ATAs that I am using to test with. If I use a mobile phone, a landline, or a digital phone to originate the call, all seems happy. If I use an ATA, it leaves just enough of the original DTMF in the audio stream to be detectable by Asterisk, but it ALSO sends an rfc2833 packet and both are detected and sent onwards! I would still be interested in any ways to improve this! Thanks, Steve On 5/3/07, Steve Davies [EMAIL PROTECTED] wrote: This is very interesting. I am now getting this double-digit behaviour occasionally, and only on IAX channels (so far). Did anyone come up with a solution or a way to improve matters? The scenario where I get this is: PSTN - Provider - IAX - Gateway - IAX - Customer So I will go and do some debugging to see where the extra packets are generated. Perhaps a simple change to prevent DTMF audio-tone detection on a channel that supports RFC or out-of-band DTMF transmission? Or perhaps some sort of DTMF debounce mechanism? Thanks, Steve On 3/6/07, Remi Quezada [EMAIL PROTECTED] wrote: Any ideas as to how I can fix this issue? Thanks Remi Remi Quezada wrote: Ok that makes sense, but I'm still getting double digits. It seems to me that the DTMF digit is getting detected too late. When the digit is pressed it seems like asterisk is passing the DTMF digit for a fraction of a second through the audio path and then sends the digit for however long your toneduration is set for. I can hear this happening when I dial the digits myself, I hear some kind sound being cut off for a fraction of a second and then hear the DTMF tone pass. So I guess this is why sometimes some answer machines are detecting double digits. Russell Bryant wrote: Remi Quezada wrote: I have two asterisk servers one is connected to the PSTN and the other one is connected to SIP users. The two servers connect with each other using IAX. When I have an incoming call from PSTN to the asterisk servers and have a forward to go back out to the PSTN the two IAX channel bridge together. Now every time I dial a DTMF digit, the asterisk is sending two DTMF digits. I enable debugging for iax and I do see it sending the DTMF digits two times. Here is what I see: The IAX debug that you show below only shows one of each digit. For each one, it shows Receiving the digit from one leg of the call, and then transmitting it out the other. I have spaced out your debug to separate each digit. Each one shows ... - digit - ACK - - digit --- ACK -- which is exactly what is supposed to happen. ___ --Bandwidth 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] using Playback() to play a random sound file
I have accomplished a similar outcome that what you are mentioning, but I use Music on Hold rather than Playback(). Using MOH was a very simple solution, although I do not know if it is specifically what you are looking for. This solution allows you to set a time limit on playback as well as continue to dial extensions. We use the /var/lib/asterisk/sounds directory and listen to it randomly - can be quite hilarious at times. To do this, we put a symlink to the sounds directory in the mohmp3 directory and enabled random play. When I want to listen to it use the following: -- [sound-player] exten = s,1,Answer exten = s,2,SetMusicOnHold(sounds) ;Name of Music context exten = s,3,WaitExten(3600|m) ;Wait (for an hour) while listing to the MOH exten = s,4,Goto(sound-player,s,2) ;After timeout, start the next hour ;OPTIONAL - Dialing 1 Skips to next audio file exten = 1,1,SetMusicOnHold(default) ;Change MOH context to drop current stream (Sometimes I would be placed back on the same music stream, this seems to reset that) exten = 1,n,WaitExten(.1|m) ;Play that stream for 1/10 of a second exten = 1,n,SetMusicOnHold(sounds) ;Reset to what you really want to hear exten = 1,n,Goto(custom-music-player,s,3) ;Go to play the MOH sounds -- This is actually part of an application that we use as an internal radio. Very simple in concept and function, but it randomly plays a directory full of sound files with the option to skip ahead to the next file. I use variables instead of static contexts though. I hope this helps you accomplish what you are trying to do. -- Thanks, Brandon Comouche -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Austad Sent: Tuesday, May 01, 2007 9:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] using Playback() to play a random sound file I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to select a file to play randomly. Is there any way to do this? ~jay ___ --Bandwidth 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] Linksys SPA3012 inbound FXO problems
Hello list, hope someone can help me - I'm going crazy using the FXO port a SPA3012. I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that is, once it detects a call, it should simply send it over to the local Asterisk server. No intelligent routing, PIN, anything else I configured it like this: PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: yes PSTN Caller Auth Method: none PSTN Ring Thru Line 1: no PSTN Caller Default DP: 1 Then I configured the dialplan #1 as: Dial Plan 1: (S0:@gw1) And I configured gateway 1 as: Gateway Accounts Gateway 1: my.asterisk.server GW1 NAT Mapping Enable: no GW1 Auth ID: --my-sip-login-- GW1 Password: --my-sip-password-- But it seems to simply ignore incoming calls at all Anybody's got a pointer to get me started? Thanks in advance, l. -- Home of QueueMetrics - http://queuemetrics.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] IAX Trunk
Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IAX Trunk
Yes it is. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronaldo Sent: Thursday, 3 May 2007 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] IAX Trunk Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth 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] IAX Trunk
Yes it is. On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
Yes it is On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Yes, Asterisk will do the conversion from SIP to IAX and back again if necessary. ___ --Bandwidth 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: USB T1/E1 Interface?
Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Just for fun. I'm a telecom geek and having a USB T1 interface would be a fun toy to tinker with. Besides, it might lead to some useful products. -MC ___ --Bandwidth 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] UK - 2 port ISDN2e cards ...
Gordon Henderson wrote: On Thu, 3 May 2007, Per Jessen wrote: (aren't you guys getting rid of ISDN anyway? :-) H... Some people would like to think so, but it's going to be here for a long time yet! BT have/are dumping the consumer versions of ISDN2 - home highway which went a while back, but business highway is going soon if it's not already gone, which is a real shame as they had almost all the functionality a small business needed for less price than the full ISDN2e... Another revenue generating feature! -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
Hi Dean, Can you suggest me any documentation about using IAX trunking? Thank you. Ronaldo. Dean Collins wrote: Yes it is. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ronaldo Sent: Thursday, 3 May 2007 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] IAX Trunk Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth 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] IP Phone Provisioning Tool by voip.com.sg - xml generation
Mats Karlsson wrote: Take a look here: http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html Ugh. This is a Win32 app, isn't it? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
Hi Bruce, Can you suggest me any documentation about using IAX truking? Thank you. Ronaldo. Bruce Reeves wrote: Yes it is. On 5/3/07, *Ronaldo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by 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] Linksys SPA3012 inbound FXO problems
I've managed to configure a SPA3012 to do that a few days ago. I remamber using something like S0:[EMAIL PROTECTED] for the #1 dial plan. Unfortunately I no longer have access to the SPA because I shiped it to an co-worker and this co-worker didn't manage to install it yet. I also remamber an odd thing: the extension really needs to exist in the correct context, it doesn't fall back to the s extension and there's no worning on the CLI ither! Also an googling tip: most configuration for the SPA3012 is the same as that for SIPURA 3000, so google for that too. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lenz Sent: Thursday, May 03, 2007 6:57 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Linksys SPA3012 inbound FXO problems Hello list, hope someone can help me - I'm going crazy using the FXO port a SPA3012. I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that is, once it detects a call, it should simply send it over to the local Asterisk server. No intelligent routing, PIN, anything else I configured it like this: PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: yes PSTN Caller Auth Method: none PSTN Ring Thru Line 1: no PSTN Caller Default DP: 1 Then I configured the dialplan #1 as: Dial Plan 1: (S0:@gw1) And I configured gateway 1 as: Gateway Accounts Gateway 1: my.asterisk.server GW1 NAT Mapping Enable: no GW1 Auth ID: --my-sip-login-- GW1 Password: --my-sip-password-- But it seems to simply ignore incoming calls at all Anybody's got a pointer to get me started? Thanks in advance, l. -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Daemontools and holidays macro
William Moore wrote: You may want to consider renaming daemontools as it is also the name of a windows program that allows you to mount CD/DVD ISOs, so there could be some confusion. Uh... This is Daniel Bernstein's 'daemontools' -- and he's not going to rename it, especially since his software predates the Windows program. -Stephen- ___ --Bandwidth 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] Daemontools and holidays macro
Vicente Aguilar wrote: El jue, 03-05-2007 a las 07:04 -0500, William Moore escribió: You may want to consider renaming daemontools as it is also the name of a windows program that allows you to mount CD/DVD ISOs, so there could be some confusion. daemontools is not the name of my scripts, but the name of a program they use to ensure Asterisk is running 100% of the time: http://cr.yp.to/daemontools.html *This* daemontools spawn a process and monitor it, re-running it if it dies. Quite useful for high-availability services. I'm surprised nobody mentioned using daemontools for Asterisk sooner. -Stephen- ___ --Bandwidth 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] Digital Phones
Jason Fuermann wrote: I've used these gateways and never experienced any of these problems. I could imagine me missing the popping noise but I do know that MWI did work just fine. What he said was that he couldn't turn stutter dialtone off, not that the MWI didn't work. Not hearing the DTMF would be a show stopper for me too. -Stephen- ___ --Bandwidth 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] Large dial plans and variables
Andreas Sikkema wrote: You're so right! I thought about having just a catchall _. extension in the dialplan and doing everything else in a real language via AGI - PHP, Perl, ... whichever you like. It would make the programming part much easier as the scope of variables is just as you expect it to be. Well, they're called macro's for a reason You guys are proposing adding functions or procedures. My first step in any macro would be to copy incoming variables, be it arguments or even asterisk defined stuff to local variables. But that is just me and my coding convention. I guess we are. I propose we add functions or procedures! Until that time though, it seems best practices are to prefix every single variable in macros, including copying ARG parameters to variables, with the name of the function, to avoid stepping on yourself. ___ --Bandwidth 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: USB T1/E1 Interface?
Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). I can well understand the idea of having USB T1 adapters since that way you can colocate 1U Asterisk systems ;-) which at least doubles you density in a rack... Frank Frank Gorgas-Waller Explido Software USA Inc. Phone +1-863-248-1195Fax +1-863-248-1155 EMail [EMAIL PROTECTED]ICQ 7733546 --QQ- We teach penguin to fly http://www.explido.us ___ --Bandwidth 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] Linksys SPA3012 inbound FXO problems
On Thu, 2007-05-03 at 17:56 +0200, lenz wrote: Hello list, hope someone can help me - I'm going crazy using the FXO port a SPA3012. I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that is, once it detects a call, it should simply send it over to the local Asterisk server. No intelligent routing, PIN, anything else I configured it like this: PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: yes PSTN Caller Auth Method: none PSTN Ring Thru Line 1: no PSTN Caller Default DP: 1 Then I configured the dialplan #1 as: Dial Plan 1: (S0:@gw1) And that's where it started to go wrong. (S0:[EMAIL PROTECTED]:5060) will do what you want. -- Dave Cotton [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] you have been kicked my this conference
Replace it with a pause sound byte. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Thursday, May 03, 2007 11:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] you have been kicked my this conference How do I stop the you have been kicked by this conference message from speaking? I first had MeetMe(conf, l) and I get the kicked message. I tried Meetme(CONF, lq) and I still get he kicked message. and it still says it. Thanks, Jerry ___ --Bandwidth 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] CDR(accountcode) empty in * 1.4.4 (for local chan)
On Thu, 2007-05-03 at 16:47 +0500, Rizwan Hisham wrote:
Hi all,
i just updated to asterisk 1.4.4 from 1.4.2. i was doing this to
forward an unanswered call in 1.4.2
exten= 1,1,Dial(SIP/123,,Ttg)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10)
exten= 1,3,Hangup
exten= 1,10,Dial(Local/2,,Ttg)
exten= 1,11,Hangup
exten= 2,1,Dial(SIP/234,,Ttg)
exten= 2,2,Hangup
All the CDR variables for the first channel (SIP/123) are fine. but
when local channel initiates, it does not copy the CDR(accountcode)
variable from the first channel (in asterisk 1.4.4), whereas it did in
1.4.2. so the CDR(accountcode) variable for local channel is empty in
1.4.4. This is a big problem for me as i have to charge the forwarded
calls also and all calls are charged based on account code. If
accountcode is empty, i cant make a decision how to charge the call.
Can anybody fix this for me or do i have to jump back to asterisk
1.4.2?
--
Regards
Rizwan Hisham
Software Engineer
Riswan--
This could easily be my fault. I've attached a fix, that I can commit to
the source, if it works for you.
Here the instructions:
1. save the attachment to a file.
2. cd to your 1.4-source/channels directory
3. patch -p0 localfix
4. cd ..
5. make
6. make install
test
If there's no differences, you still have the same problem, you'd best
restore the source to it's previous condition:
1. cd 1.4-sourcedir/channels
2. mv chan_local.c.orig chan_local.c
3. cd ..
4. make
5. make install
This patch will properly set the accountcode amaflag from the local
channel's owner at channel creation time, and therefore, the local
channels' CDR as well.
--
Steve Murphy
Software Developer
Digium
Index: chan_local.c
===
--- chan_local.c (revision 62984)
+++ chan_local.c (working copy)
@@ -594,10 +594,22 @@
{
struct ast_channel *tmp = NULL, *tmp2 = NULL;
int randnum = ast_random() 0x, fmt = 0;
+ const char *t;
+ int ama;
/* Allocate two new Asterisk channels */
- if (!(tmp = ast_channel_alloc(1, state, 0, 0, , p-exten, p-context, 0, Local/[EMAIL PROTECTED],1, p-exten, p-context, randnum))
- || !(tmp2 = ast_channel_alloc(1, AST_STATE_RING, 0, 0, , p-exten, p-context, 0, Local/[EMAIL PROTECTED],2, p-exten, p-context, randnum))) {
+ /* safe accountcode */
+ if (p-owner p-owner-accountcode)
+ t = p-owner-accountcode;
+ else
+ t = ;
+
+ if (p-owner)
+ ama = p-owner-amaflags;
+ else
+ ama = 0;
+ if (!(tmp = ast_channel_alloc(1, state, 0, 0, t, p-exten, p-context, ama, Local/[EMAIL PROTECTED],1, p-exten, p-context, randnum))
+ || !(tmp2 = ast_channel_alloc(1, AST_STATE_RING, 0, 0, t, p-exten, p-context, ama, Local/[EMAIL PROTECTED],2, p-exten, p-context, randnum))) {
if (tmp)
ast_channel_free(tmp);
if (tmp2)
smime.p7s
Description: S/MIME cryptographic signature
___
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Re: [asterisk-users] IAX Trunk
On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote: Can you suggest me any documentation about using IAX trunking? Thank you. There are examples in the iax.conf files I think, but basically just put something like [iax-toremote] type=friend username=whatever secret=somesecret auth=plaintext host=somewhere.com peercontext=some-context qualify=yes trunk=yes then you dial with Dial(iax2/iax-toremote/number) 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/Mac 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
Re: [asterisk-users] IAX Trunk
On Thu, 3 May 2007, Ronaldo wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Absolutely. Have a look at this: http://astrecipes.net/index.php?n=204 Gordon ___ --Bandwidth 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] [Announce] Web-MeetMe V3.0.1 released
Ondrej wrote: I finally got some time to test the SVN branches and here are my comments: Cool. One thing that does not work for sure - I had some problems to terminate the running conference from within the web page - I just clicked the button and nothing happened. This is likely a manager.conf security issue, but it could be a problem in the php code. I just tested branches/3.0 and trunk against 1.4.1 and it worked as expected. If you set core verbose to 10 and click on 'End Now' the console should display a message like this: app_meetme.c:941 meetme_cmd: Cmdline: 41251|k|1 At this point this would be a topic better suited for the support forums on SF. I browsed the php sources trying to understand and from what I see, the End Now button does nothing else than kicking all attendees - not exactly what I would expect. I would expect this action to terminate the conference immediately so, it could be seen in the Past conferences list. Also, some javascript popup dialog confirming this action would be nice. The same is valid for the Extend button - it works, but from the user prospective, nothing happens - I would expect some dialog box like The conference # has been extended by 10 minutes. This is the only missing piece, I would say - thanks :-) Oh! Those are great ideas and fairly easy to add. I'm about to be offline for two weeks, and need to get updated releases of 2.X and 3.X out before I go. Your ideas are now on the ToDo list and I'll try to get them integrated and released by mid-June. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Autologoff
I am having an issue with the autologoff fuction in agents.conf I am running Asterisk BE and I am testing with agent 1656. I log in, and then make a call into the queue. The agent's phone rings, and if I answer it, all's fine/ I am trying to have this agent automatically be logged off if he does not answer the queue callback within 5 seconds, however the agents extension keeps ringing until the call eventually goes to the extension's voice mail, which I am also trying to avoid. Here is my agents.conf [general] persistentagents=yes [agents] autologoff=5 multiplelogin=no recordagencalls=yes monitor-join=yes createlink=yes updatecdr=yes musiconhold=default recordformat=wav49 urlprefix=http://64.211.222.226/calls/ savecallsin=/var/www/html/calls agent = 1650,1650,Tareq Tujjar agent = 1656,1656,Ed Nuñez Here is my queues.conf [general] persistentmembers=yes [noi-cust-serv-spanish] strategy = leastrecent announce-frequency = 90 announce-holdtime = yes announce-round-seconds = 10 timeout=180 monitor-format=wav49 monitor-join=yes joinwhenempty = strict leavewhenempty = yes musiconhold = default eventwhencalled = yes queue-youarenext = queue-youarenext; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting; (calls waiting.) queue-holdtime = queue-holdtime; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou; (Thank you for your patience.) queue-lessthan = queue-less-than ; (less than) member = Agent/1656 image001.png___ --Bandwidth 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] using Playback() to play a random sound file
Steve Edwards wrote: On Tue, 1 May 2007, Jay Austad wrote: I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to select a file to play randomly. Is there any way to do this? I do this with an AGI. On Wed, 2 May 2007, dave cantera wrote: here is a way that I solved a similar problem... have a shell script that runs and indexes all the files in the directory into an ascii flat file with a format of filename 0001 directory/tt-weasels 0002 directory/tt-monkeys in your dialplan use the rand() to pick a number, pass it to the shell script as an arg[], then the shells script grep()'s and cut()'s the filename puts it in a db varaible, the dialplan picks it up and plays it... as you can see, I haven't done it yet :) but, in theory it works... you could skip the dialplan rand() and just use linux rand based on the minutes or seconds value for current time... you don't have to zero fill the index either, I seem to like nicely formated files, they are easier for humans to read. daveC Sounds like a lot of effort to avoid writing an AGI. If you have the skills to write the script described above, you have the skills to write an AGI -- you can write AGI's in shell scripts, btw. AGI's accept stuff from Asterisk on stdin and send stuff back to Asterisk on stdout -- very simple and elegant actually. Take your script and rewrite the reading arguments bits to read from stdin and change the write db bits to write to stdout (set a channel variable) and you have an AGI and a much cleaner dialplan. I write AGI's in C for speed and flexibility. No interpreter (bash, perl, php, etc.) to fire up, full access to anything you want to do. In C, I call ftw() (ftw - traverse (walk) a file tree). If I get more than 1 file, I choose one randomly. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth 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] Called party identification - where to take calledname?
Yehavi wrote: I am trying to apply the called party identification patch (patch 8824) and managed to make it work with a static data. Where do I take the name of the called person (the equivalent of CALLERID, but the other way...)? Short answer is that you cannot. Longer answer is that it is possible, but requires new functionality to be added to the core and a new API call be added that can check if the called party is a local endpoint and retrieve the caller-id values. At least that was what I found when working on the patch. If anyone knows a way to lookup a peer/friend from the dialplan and collect such details, it would be possible to use the existing patch without any more changes in the core. BTW, one note to the above patch: To make it work the device should have the parameter sendrpid set to true. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-Polycom HELLLLPPP!!!!
PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to find that he is talking to User1 User2 is stuck in music-on-hold. FOREVER! The other two phones work exactly as they should using the # key Using the # key on the Polycom only allow the dialing of 1 number before Alice announces That there is no such extension. HELP Thanks in advance 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] Semi-OT: useful things to do with XML browsers inphones
Chris wrote: It seems that more and more phones these days are coming with XML mini-browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. So I'm open to suggestions. What useful applications are you developing for these mini-browsers? What sort of things do your customers want to use on them? I've been planning to write to app for joining scheduled conferences. It would be bundled with the Web-MeetMe suite. Users of the app would see a list of conferences scheduled for the current time and have one-button access to the conference (assuming no PINs) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
The wiki has a decent page about it. http://www.voip-info.org/wiki-IAX What you are trying to setup sounds simple enoug, you mainly will have an extension or pattern match that executes a dial command from box A to box B and passes the remote extension. On 5/3/07, Ronaldo [EMAIL PROTECTED] wrote: Hi Bruce, Can you suggest me any documentation about using IAX truking? Thank you. Ronaldo. Bruce Reeves wrote: Yes it is. On 5/3/07, *Ronaldo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by 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 -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zttranscode crashes server
On Thu, May 03, 2007 at 10:52:51AM -0400, Forrest Beck wrote: So is anyone not using the zaptel init script to load modules? The fix I mentioned was about unloading rather than about loading. Anyone using modules.conf? How an I load them at boot without using the init script? Do I just remove --ignore-install from modprobe? If you use the init.d script, you don;t need any special install commands: just use the init.d script, and it will run ztcfg when everything is loaded. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth 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] chan_sip seems to be hanging
try soft hangup sip channel name On 02/05/07, Ken Williams [EMAIL PROTECTED] wrote: I posted about this problem last week and thought it was a combination of SIP/ZAP causing issues in Asterisk. Since then I've realized it's only the SIP channel that's hanging. When this happens a call can still come in and hit the IVR, but no one can connect to the server from a SIP client. I tried reloading chan_sip.so today when this occurred, and I tried unloading chan_sip.so but was told the channel was in use. How can I clear SIP connections? With ZAP channels I can use ZAP DESTROY CHANNEL, but I don't see the equivalent for SIP. Any suggestions for tracking down what's causing SIP to hang? My only option as it stands is to shutdown asterisk restart it, I included a piece of the log last week and am willing to do so again if needed. If I can see which SIP channels the server thinks are open when the channel hangs I'm hoping this will allow me to find if it's a common phone or perhaps some dialplan logic gone bad. Thanks, Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IAX Trunk
Good luck. Try these. http://www.voip-info.org/wiki-IAX http://www.voip-info.org/wiki-IAX+versus+SIP http://www.voip-info.org/wiki/view/IAX+encryption http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo Sent: Thursday, May 03, 2007 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Trunk Hi Bruce, Can you suggest me any documentation about using IAX truking? Thank you. Ronaldo. Bruce Reeves wrote: Yes it is. On 5/3/07, *Ronaldo* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, Is it possible to have something like this: SoftPhone -(SIP)- Asterisk -(IAX trunk)- Asterisk -(SIP)- SoftPhone I want a IAX trunk between two asterisks and on each tip I have SIP clients that need to talk to each other. Thansk. Ronaldo ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by 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: USB T1/E1 Interface?
On Thu, 3 May 2007, [EMAIL PROTECTED] wrote: Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). I can well understand the idea of having USB T1 adapters since that way you can colocate 1U Asterisk systems ;-) which at least doubles you density in a rack... ) Impress the hell out of a client -- Our PBX smoked and this guy is running our telecoms on his laptop! ) Inline/passthrough diagnosis and monitoring -- kind of like a network sniffer. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth 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: USB T1/E1 Interface?
http://www.gl.com/laptopt1.html Jorge Michael Collins wrote: Why? There used to be a saying 'usb is for mice, firewire is for men', though USB has grown a bit in bandwidth since then, it is still not very well suited for a high sustained bandwidth. NOw T1/E1 is not that big, I suspect a lack of demand. Havng a E1 termintae in your laptop is quite useless, and a server usually has plenty of slots (if not, buy a bigger server ;-). Just for fun. I'm a telecom geek and having a USB T1 interface would be a fun toy to tinker with. Besides, it might lead to some useful products. -MC ___ --Bandwidth 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] ast_parse_allow_disallow: Cannot disallow unknown format ''
Hello, everyone. I've installed asterisk SVN-branch-1.4-r62942 and every time I reload asterisk I get this in CLI: -- Reloading module 'app_playback.so' (Sound File Playback Application) [May 3 20:04:26] NOTICE[13892]: app_playback.c:455 reload: Reloading say.conf == Parsing '/etc/asterisk/say.conf': Found [May 3 20:04:26] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow: Cannot disallow unknown format '' [May 3 20:04:26] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow: Cannot disallow unknown format '' [May 3 20:04:26] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow: Cannot disallow unknown format '' [May 3 20:04:27] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow: Cannot disallow unknown format '' Is that something to worry about? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation
On 5/3/07, Stephen Bosch [EMAIL PROTECTED] wrote: Ugh. This is a Win32 app, isn't it? Yup. ___ --Bandwidth 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: USB T1/E1 Interface?
I can well understand the idea of having USB T1 adapters since that way you can colocate 1U Asterisk systems ;-) which at least doubles you density in a rack... Frank I'm glad I asked the question! I was just thinking to myself that it would be cool to have a USB T1 adapter so that I could tinker, but you guys have already come up with several real-world applications! I think I will research this some more and let you all know if anything interesting pops up. -MC ___ --Bandwidth 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] Linksys SPA3012 inbound FXO problems
Thanks a lot, that did the trick! Wish there was an half-decent manual on the site at least :-( l. In data Thu, 03 May 2007 18:34:34 +0200, Dave Cotton [EMAIL PROTECTED] ha scritto: On Thu, 2007-05-03 at 17:56 +0200, lenz wrote: Hello list, hope someone can help me - I'm going crazy using the FXO port a SPA3012. I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that is, once it detects a call, it should simply send it over to the local Asterisk server. No intelligent routing, PIN, anything else I configured it like this: PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: yes PSTN Caller Auth Method: none PSTN Ring Thru Line 1: no PSTN Caller Default DP: 1 Then I configured the dialplan #1 as: Dial Plan 1: (S0:@gw1) And that's where it started to go wrong. (S0:[EMAIL PROTECTED]:5060) will do what you want. -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Trunk
OK Steve, Just one more question. Using this configuration can I make more than one call at the same time? Thanks. Steve Kennedy wrote: On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote: Can you suggest me any documentation about using IAX trunking? Thank you. There are examples in the iax.conf files I think, but basically just put something like [iax-toremote] type=friend username=whatever secret=somesecret auth=plaintext host=somewhere.com peercontext=some-context qualify=yes trunk=yes then you dial with Dial(iax2/iax-toremote/number) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linseed
http://www.linuxdevices.com/news/NS3013985136.html Ok so who's going to be the first to install Asterisk on it? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation image001.gif___ --Bandwidth 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] Strange noise - Polycom
I'm not sure if this is a problem with our polycom 501 phones or with a setting in asterisk. When you set the forward option on the phone and have it point to an outside number (a cell phone) we see the following problem... The call does forward, but while its doing so and while its ringing, you hear this irritating loud pulsing sound. When we call from a non-polycom phone and dial into the phone with the forward, this noise isn't there. I couldn't find a polycom mailing list, otherwise I'd try this question in there as well. Has anyone ever seen a problem like this? Rob ___ --Bandwidth 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] Wildcard TE410P problem
Hi Alexander and the list, Have you well checked your E1 cable ? Sometime, you must use a crossed E1 cable (not an Ethernet one)... Check also without the crc check. How is your zapata.conf file ? Have you checked with a loop (crossed E1 cable) between two spans (one in TE the second in NT, of course) ? Best Regards, Francois BERGERET, France. ___ --Bandwidth 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] Wildcard TE410P problem
Autocorrection mode : pri_cpe / pri_net rather than TE / NT ;-) -Message d'origine- De : Francois BERGERET [mailto:[EMAIL PROTECTED] De la part de '[EMAIL PROTECTED]' Envoyé : jeudi 3 mai 2007 21:03 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : RE : [asterisk-users] Wildcard TE410P problem Hi Alexander and the list, Have you well checked your E1 cable ? Sometime, you must use a crossed E1 cable (not an Ethernet one)... Check also without the crc check. How is your zapata.conf file ? Have you checked with a loop (crossed E1 cable) between two spans (one in TE the second in NT, of course) ? Best Regards, Francois BERGERET, France. ___ --Bandwidth 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] FXO recommendation
Hi all, With the gamut of FXO cards out there, I'm looking for a recommendation for home use. I have a nicely working Asterisk 1.4 system that just requires an FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have kicked the bucket. Any suggestions would be greatly appreciated. Regards Kyle -- Kyle Gordon [EMAIL PROTECTED] http://lodge.glasgownet.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!
Jim, What happens in your first senario is an attended transfer, after User1 and 3 have initiadted their call, User1 should press transfer again to complete the transfer. At which point User1 will be disconnected and Users 2 3 will talk. The second issue is the limit of digits and is likely due to a very short timeout in features.conf, check the entry transferdigittimeout. On 5/3/07, Jim Suber [EMAIL PROTECTED] wrote: PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to find that he is talking to User1 User2 is stuck in music-on-hold. FOREVER! The other two phones work exactly as they should using the # key Using the # key on the Polycom only allow the dialing of 1 number before Alice announces That there is no such extension. HELP Thanks in advance 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 -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!
On Thu, 3 May 2007, Jim Suber wrote: PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) Yes ... User3 picks up the phone to answer User2 only to find that he is talking to User1 This is as expected. It's an ATTENDED transfer - User 2 (external?) calls in. U1 answers. User 1 has placed User 2 on hold User 1 calls U3 User 1 speaks to U3, asks U3 if they want to accept the call from U2. and if yes, then U1 has to do whatever is neccessary to transfer U2 to U1. If no, then U1 has to retrieve U2 and say sorry they are not at their desk (or whatever) Maybe you're confusing it wuth UNattended transfer? I've no knowlege of the Polycoms, but on the Grandstreams, for unattended xfer, you hit the transfer button, dial the extension, then hang up - U2 gets ringing and U1's phone rings. (Attended transfer is slightly more complicated) Gordon ___ --Bandwidth 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] Semi-OT: useful things to do with XML browsersinphones
Neat, not something I would be interested in but I can certainly see how people would use that. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dan Austin Sent: Thursday, 3 May 2007 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Semi-OT: useful things to do with XML browsersinphones Chris wrote: It seems that more and more phones these days are coming with XML mini-browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. So I'm open to suggestions. What useful applications are you developing for these mini-browsers? What sort of things do your customers want to use on them? I've been planning to write to app for joining scheduled conferences. It would be bundled with the Web-MeetMe suite. Users of the app would see a list of conferences scheduled for the current time and have one-button access to the conference (assuming no PINs) ___ --Bandwidth 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-Polycom HELLLLPPP!!!!
Whoa. Calm down. Jim Suber wrote: PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to find that he is talking to User1 User2 is stuck in music-on-hold. FOREVER! That's because you didn't finish transferring User2. EVER! I can tell that you haven't bothered to read the Polycom user's guide, which is a bit annoying, but I will do you this one favour. Attended transfers on the Polycom phones work like this: 1. User A calls you. 2. You answer the phone. 3. You press 'Transfer' -- at this point User A hears music -- and dial User B, then press 'Send' 4. You will hear User B's line ring. YOU AREN'T DONE YET. 5. Now press 'Transfer' again. NOW you're done. Hang up. Note -- you are not done with a transfer procedure until your phone is in the idle state again. That is, once you've transferred a call, the call will disappear from the phone's display. If you still see User A's call on your phone's display, something is wrong. Now that I have helped you, go read the Polycom documentation. The list likes to help people who have tried to help themselves first. -Stephen- ___ --Bandwidth 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-Polycom HELLLLPPP!!!!
Jim Suber wrote: PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom I'm not sure about the 430, but on the 501, you'd do a transfer, talk to user3, if user3 wants the call, you'd press transfer again to complete the call. If not, you'd press the resume to get the call back. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth 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-Polycom HELLLLPPP!!!!
Isn't that the function of an attended transfer? User3 hears User1 to see if they want to take the call or not. User1 should then hit the transfer key again to finalize the call. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Suber Sent: Thursday, May 03, 2007 12:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk-Polycom HEPPP PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to find that he is talking to User1 User2 is stuck in music-on-hold. FOREVER! The other two phones work exactly as they should using the # key Using the # key on the Polycom only allow the dialing of 1 number before Alice announces That there is no such extension. HELP Thanks in advance 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
