Re: [asterisk-users] IAX2/SIP gateway for Belgium and western Europe
On 19/01/07, Jan Dewerchin [EMAIL PROTECTED] wrote: Dear all, I'm not sure if this is the correct place to put it, but can I announce you the possibility of using a new, lost-cost trunk for Belgium and western Europe ? Maybe it's a shameless commercial plug, but have if you don't know it exists, how can you all benefit from this ? asterisk-biz is the correct place. This isn't. Peter -- Peter Bowyer Email: [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] NAT solutions
On Thu, 18 Jan 2007, Voip Asterisk wrote: I know that NAT is something no one really likes to talk about, but does anyone know how work with it elegantly? There are many providers which deal with it on a daily basis in fact they cater to it, is this possible to do with asterisk or does it require other exotic setups? I even know of a provider which uses asterisk with many different types of devices, and they handle all NAT config on their end even to the point of deciding to stay in the media stream or not (ie when two endpoints are behind NAT you almost have to stay in the media stream unless you got it figured out like skype does). What is the best way to work with NAT, and build a production system? I have successfully installed * boxes behind NAT firewalls and had client devices (SIP phones) talk to it, with themselves being behind NAT firewalls without doing anything overly special, or using specialised appliances, SIP gateways and so on. If you only have one * box behind the NAT gateway then I don't really see a big issue with it to be honest. Port-forward on the firewall/router device (5060 and 1 through 2) to the * device, and use STUN on the client device to help it get through it's local NAT firewall/router. I have seen issues with overly clever NAT devices - Junipers for example. They have a SIP helper application, but I reckon it's broken - when we turned it off and reverted to basic port forwarding everything was sweet. You do need additional runes in sip.conf: nat=yes externip=1.2.3.4 localnet=192.168.2.0/24 which makes a big difference! (asterisk 1.2.x) It doesn't solve the data traffic routing though - the * box does have to route traffic between 2 external SIP devices, alas. 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
[asterisk-users] mysterious SIP packets to Cogent
In my SOHO setting even when nobody is using the phone my firewall drops outgoing packets from the asterisk box to a Cogent server, din't find naything through Google about it: (out: eth0 xxx.xxx.xxx.xxx.:2129 - 66.250.40.33:24441 UDP len:193 ttl:64). Anyone know what this traffic is supposed to be good for? Greetings --AvH ___ --Bandwidth 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] Voice Recognition
Hi all, Does anyone know if Asterisk or any available 3rd party add-on for it support voice recognition (not speech recognition) - task of recognizing people from their voices? Thanks, Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Recognition
My voice is my passport; verify me. ;) I don't think you'll get reliable results with 8khz sample rates. The highest frequency wave you can achieve is a 4khz square wave. Anyway, i don't think if such software exists ;) Julian J. M. On 1/19/07, Asterisk [EMAIL PROTECTED] wrote: Hi all, Does anyone know if Asterisk or any available 3rd party add-on for it support voice recognition (not speech recognition) - task of recognizing people from their voices? Thanks, Alex ___ --Bandwidth 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] How to limit IAX calls
Take a look on: Dialplan applications: GetGroupMatchCount([EMAIL PROTECTED]) SetGroup([EMAIL PROTECTED]) Using this two applications you can deploy a max calls control inside your dialplan! check this too: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup Hope it helps On 1/19/07, Barzilai Spinak [EMAIL PROTECTED] wrote: The SIP channels have a call-limit parameter (which is badly documented and I haven't tested yet) How can I have the same behaviour for IAX channels? I can't see anything related to it. Ah, I'm using Asterisk 1.2.13... maybe there is something in the 1.4 versions... but I can't change to 1.4 right now because of MFC/R2 BarZ ___ --Bandwidth 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] NAT solutions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Gordon Henderson wrote: If you only have one * box behind the NAT gateway then I don't really see a big issue with it to be honest. Port-forward on the firewall/router device (5060 and 1 through 2) to the * device, and use STUN on the client device to help it get through it's local NAT firewall/router. I use the same strategy and it works just fine, however, I did have an issue, here's the scenario: SIP Ext 1 +---+ |--- NAT -- Internet -- NAT -- Asterisk SIP Ext 2 +---+ Each of the SIP extensions work individually, but if I try to use both of them, only the first one registers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFsLr42QVs8jsa1mQRAgUKAJ9q7Fr9wIEGXOIXPN8VCgWyCPPHlwCgopG2 DJIKK8NRsAzXbj/MrFtazks= =UPC0 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme ${DATETIME} variable update
Hi i am experiencing this problem:
MEETME_RECORDINGFILE=/data/asterisk_data/_${DATETIME}_CONFERENCE
exten = ,1,MeetMe(666|1Arxq)
exten = 9998,1,MeetMe(666|1Axq)
exten = 9997,1,MeetMe(666|1xq)
I make a conference between 3 person dialing
A dials
B dials 9998
C dials 9997
all works fine but the datetime won't be updated, it still remain for
example 13:40 until i do a complete restart of asterisk.
where can be the problem?
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[asterisk-users] pickup call out of menu
Is it possible to pickup a caller, who is in the menus somewhere, for instance he may be lost in the telemarketer torture script? Just like it is possible to pick up a call on a ringing phone. Leif ___ --Bandwidth 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] direct transfer in features
I have some siemens wireless ip-phones. There is no problem entering ** which I have configured in features.conf to be transfer. But then it is difficult to enter the extension, because one have to wait the right amount of time before entering the extension. Because we only have few extensions, is it possible to have each transfer-option as a separate feature in features.conf So can I hardwire **1 to transfer to extension 11, **2 to extension 12 *** to park etc. ? Leif ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Phone CID
Rob Schall wrote: This might sound like an odd question but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But... our old phone system set the caller id on the senders phone to show who they called. Example... If Sally calls Jim, then Sally's phone should just say 1001, it should say Jim 1001. Any know if this is possible. Our old PBX did this, and the bosses were curious if this is possible. Thanks, Rob I have tried over and over to figure out how to do this and it doesn't seem possible at the moment. I know this can be done with chan_sccp and maybe even chan_skinny (haven't tried that in a few years), but you'd need Cisco phones to do it. Is this something on anyone's To-Do list? Thanks, Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to limit IAX calls
A demonstration:
exten = _X.,1,Set(GROUP()=${CALLERID(num))
exten = _X.,n,Set(CDR(AccountCode)=${CALLERID(num))
exten = _X.,n,GotoIf($[${GROUP_COUNT(${CALLERID(num))} 2]?103)
exten = _X.,n,Macro(trunk,${EXTEN},residential)
exten = _X.,n,Hangup
exten = _X.,103,Playback(allison7/all-circuits-busy-now)
exten = _X.,n,Hangup
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
Mouta
Sent: Friday, January 19, 2007 6:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to limit IAX calls
Take a look on:
Dialplan applications:
GetGroupMatchCount([EMAIL PROTECTED])
SetGroup([EMAIL PROTECTED])
Using this two applications you can deploy a max calls control inside
your dialplan!
check this too: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup
Hope it helps
On 1/19/07, Barzilai Spinak [EMAIL PROTECTED] wrote:
The SIP channels have a call-limit parameter (which is badly
documented and I haven't tested yet)
How can I have the same behaviour for IAX channels? I can't see anything
related to it.
Ah, I'm using Asterisk 1.2.13... maybe there is something in the 1.4
versions... but I can't change to 1.4 right now because of MFC/R2
BarZ
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RE: [asterisk-users] Voice Recognition
Hi Alex, I've spoken to some commercial (read 'large company') RD people who were messing around with telephony based voice recognitionnot great results and project was abandoned (basically the confirmation threshold was going to have to be set so low it wasn't worth it). Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Friday, 19 January 2007 6:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Voice Recognition Hi all, Does anyone know if Asterisk or any available 3rd party add-on for it support voice recognition (not speech recognition) - task of recognizing people from their voices? Thanks, Alex ___ --Bandwidth 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] DND - message
Thanks very much for your response Andrew... Andrew Joakimsen wrote: Well now that I look into it, if you disable the call waiting the response is 486 busy here. If you use the DND it is the same response, so there's no way to do phone-side DND and correctly report the voicemail state. But the Aastra phones do support the selection, which IMO should be 603 decline when the phone is in DND. I suppose the RFC isnt exactly clear but I don't think Asterisk's internal call handeling should be considered another endpoint ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: meetme ${DATETIME} variable update
On 1/19/07, nik600 [EMAIL PROTECTED] wrote:
Hi i am experiencing this problem:
MEETME_RECORDINGFILE=/data/asterisk_data/_${DATETIME}_CONFERENCE
exten = ,1,MeetMe(666|1Arxq)
exten = 9998,1,MeetMe(666|1Axq)
exten = 9997,1,MeetMe(666|1xq)
I make a conference between 3 person dialing
A dials
B dials 9998
C dials 9997
all works fine but the datetime won't be updated, it still remain for
example 13:40 until i do a complete restart of asterisk.
where can be the problem?
it seems that i've fixed using
Set(MEETME_RECORDINGFILE=/data/asterisk_data/_${DATETIME}_CONFERENCE)
before each meetme
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Re: [asterisk-users] function call out of AGI script
The documentation says that the exec() command exists to execute
applications, not functions.
How would I convert an dialplan extension like
exten = 11,1,Set(count_naptr=${ENUMLOOKUP(4961369993473,ALL,c)})
into an exec call like
$AGI-exec($app, $options) ?
On 18/01/07, William Piper [EMAIL PROTECTED] wrote:
It is called exec.
http://www.google.com/search?hl=enq=asterisk+agi+exec
On 1/18/07, Thomas Hecker [EMAIL PROTECTED] wrote:
Hi everyone,
Is it possible to call an asterisk function out an AGI script? How do I
do this?
Thank you,
Thomas
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[asterisk-users] CPU Bandwidth Consumption
Is the Asterisk processing and mixing of SIP channels into a single call (simple/minimum, no transcoding etc) calculated in integer or floating point instructions? How much CPU bandwidth is used per call leg, in either MIPS or MFLOPS? How about the G.729 codec, or other codecs: MIPS/MFLOPS? Any ideas how efficient is the Asterisk/x86 code compared to the maximum in the algorithm, that either SW optimization or porting to a more efficient processor (or both) could produce? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay in Call Distribution using the Queue Application
On Jan 15, 2007, at 2:22 PM, [EMAIL PROTECTED] wrote: Hello all, we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent. For example there are 10 callers in the queue, an Agent is finishing a call and it takes up to 30 seconds before his phone rings again. We're already set the wrapuptime parameter in queues.conf to 0, for my point of view an agent phone that becomes available again should ring immediately after hanging up a call. Does anybody know if there are any known issues or restrictions in the queue application in version 1.2.12.1? You may be running into the limitation in Asterisk 1.2 (It's fixed in 1.4, I think double check that) in how the queues distribute calls. Basically, the queue can only distribute one call at a time, so if you have two agents, both available, and two calls in the queue, asterisk will send call #1 to agent #1 first. Once that call is connected, Asterisk will then send call #2 to agent #2. In other words, until asterisk distributes the first call, it can't distribute any other calls waiting in line. One nasty side effect of this is that an agent who fails to log out and leaves their desk will add 30 seconds or so (the amount of time their phone rings before the queue gives up and tries the next agent) wait time to all of the calls waiting in the queue. Tom ___ --Bandwidth 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] Announce option for meetme - is it used?
Announce option for meetme - is it used? It makes a caller record their name, but I do not see where this name recording is ever used. Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Fax. 248-836-5101 www.hirotecamerica.com Board member of www.glimasoutheast.org Our company name has changed to HIROTEC AMERICA www.hirotecamerica.com Please update any contact info with my new email address [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] Sip Phone CID
check out rpid Mark Johnson wrote: Rob Schall wrote: This might sound like an odd question but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But... our old phone system set the caller id on the senders phone to show who they called. Example... If Sally calls Jim, then Sally's phone should just say 1001, it should say Jim 1001. Any know if this is possible. Our old PBX did this, and the bosses were curious if this is possible. Thanks, Rob I have tried over and over to figure out how to do this and it doesn't seem possible at the moment. I know this can be done with chan_sccp and maybe even chan_skinny (haven't tried that in a few years), but you'd need Cisco phones to do it. Is this something on anyone's To-Do list? Thanks, Mark ___ --Bandwidth 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] Announce option for meetme - is it used?
It is played to the conference/meetme room prior to the user entering, at least in our setup it works. The caller does not here their own announcement. On 1/19/07, BerkHolz, Steven [EMAIL PROTECTED] wrote: Announce option for meetme - is it used? It makes a caller record their name, but I do not see where this name recording is ever used. Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Fax. 248-836-5101 www.hirotecamerica.com Board member of www.glimasoutheast.org Our company name has changed to HIROTEC AMERICA www.hirotecamerica.com Please update any contact info with my new email address [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 -- Bruce 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] Sip Phone CID
I set both the trustrpid and sendrpid to yes, but the calling phone still doesn't show the caller id of the person they are calling. Jason Fuermann wrote: check out rpid Mark Johnson wrote: Rob Schall wrote: This might sound like an odd question but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But... our old phone system set the caller id on the senders phone to show who they called. Example... If Sally calls Jim, then Sally's phone should just say 1001, it should say Jim 1001. Any know if this is possible. Our old PBX did this, and the bosses were curious if this is possible. Thanks, Rob I have tried over and over to figure out how to do this and it doesn't seem possible at the moment. I know this can be done with chan_sccp and maybe even chan_skinny (haven't tried that in a few years), but you'd need Cisco phones to do it. Is this something on anyone's To-Do list? Thanks, Mark ___ --Bandwidth 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] connecting a FXS-to-sip 4 port device to an avaya system
analog station ports = fxs analog line ports = fxo, assuming 2 wire loop start On Jan 18, 2007, at 8:26 PM, Erick Perez wrote: Thanks Jerry. Are the avaya station ports a special type ? On 1/18/07, Jerry Jones [EMAIL PROTECTED] wrote: Connect to the avaya line ports, not station ports. On Jan 18, 2007, at 10:46 AM, Erick Perez wrote: Hi, this is a signalling question: I have a 4port fxs-to-sip where i connect standard analog phones. I want to connect this device to an avaya PBX and then the device talks to asterisk via SIP. What signalling do i need the avaya to provide? FXO signalling right, like this? avayaanalog_lines_fxo_signal--(FXS side)FXStoSIPdevice(SIP side)--Asterisk thanks, -- Erick Perez ___ --Bandwidth 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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian link? find if putting in loop your own asterisk still fails, etc etc. Kind Regards On 1/18/07, Facundo Ameal [EMAIL PROTECTED] wrote: Thanks for your help, but I've already adjusted timers on the source code. I found your document a week ago and read it. Do you really think that is a matter of timers only? Greets! On 1/18/07, Moises Silva [EMAIL PROTECTED] wrote: Sometimes timers need to be adjusted on the mfcr2 source code. Sometimes is missconfiguration. Anyway, may be this document can help you out to debug the problem: http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf Kind Regards On 1/17/07, Facundo Ameal [EMAIL PROTECTED] wrote: Hi everyone! I'm having some issue trying to place calls with asterisk connected to an E1 R2 from Telmex Argentina. The other E1 port is connected to a Meridian which also uses R2 protocol. Calls sometimes fail with different error messages such as: Unicall protocol error 32773, 32772, 32769. Some other calls fail saying: Far end disconnected(cause=Destination out of order [27]) Far end disconnected(cause=User alerting, no answer [19]) Far end disconnected(cause=Switching equipment congestion [42]) Far end disconnected(cause=User busy [17]) I don't think those causes are real, because if you use another line, yo establish the call. Could it be something about timing of ABCD bits? I'm using: Asterisk 1.2.6 Zaptel 1.2.5 libmfcr2 0.0.3 libunicall 0.0.3 libsupertone 0.0.2 spandsp-0.0.3 And this is my unicall.conf: [channels] loglevel=1023 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=no echocancelwhenbridged=no echotraining=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,4,15 protocolend=cpe group=1 context=from-zaptel channel = 1-15 channel = 17-29 loglevel=0 usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no protocolclass=mfcr2 protocolvariant=ar,0,12,12 protocolend=cpe group=2 context=hacia-afuera channel = 32-46 channel = 48-60 Thanks in advance! Greets! -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Facundo Ameal. famealatgmaildotcom Linux User #395088 Share your knowledge, use free software. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Integrating asterisk with Toshiba Astrata DK380
Deat all,
I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.
Following is my setup
*Asterisk - Digium TE110P - E1 card in toshiba pbx - Toshiba PBX*
A = B
C D
Asterisk PBX and strata PBX connected using back to back E1 cross cable.
Physicall connectivity is OK. The digium TE110p
LED state green. zttool also OK.
Toshiba stata configured to make outbound call via E1 link with pressing 9
and then the out side number.
I was able to make call from soft phone to analog extension at toshiba pbx.
A== B way as shown above. But when trying to dial from
Toshiba PBX analog extension to asterisk extension, by pressing 9 the call
rejected.
In the asterisk command prompt I'm having following error message.
-- Extension '' in context 'from-pstn' from '' does not exist. Rejecting
call on channel 0/31, span 1
Is there any wrong in my setup. dial plan??, additional configuration if i
required to put please guide me.
I will be greately appreciated your feedback on this regard.
*configuration details*
*/etc/zaptel.conf*
# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
*/etc/asterisk/zapata.conf*
signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=euroisdn
;switchtype=national
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number is
in milliseconds
callerid=asreceived
overlapdial=no
pridialplan=unknown
immediate=no
;rxwink=300
callprogress=no
loadzone=au
context=from-pstn ; Points to the default context of your extensions.conf
group=2
channel=1-15
channel=17-31 ;PRI/E1 link
[trunkgroups]
trunkgroup=2,16
spanmap=1,2,1
*/etc/asterisk/extension.conf*
[from-zaptel]
exten = _X.,1,Set(DID=${EXTEN})
exten = _X.,n,Goto(s,1)
exten = s,1,NoOp(Entering from-zaptel with DID == ${DID})
; If ($did == ) { $did = s; }
exten = s,n,Set(DID=${IF($[${DID}= ]?s:${DID})})
exten = s,n,NoOp(DID is now ${DID})
exten = s,n,GotoIf($[${CHANNEL:0:3}=Zap]?zapok:notzap)
exten = s,n(notzap),Goto(ext-did,${DID},1)
; If there's no ext-did,s,1, that means there's not a no did/no cid route.
Hangup.
exten = s,n,Macro(hangup)
exten = s,n(zapok),NoOp(Is a Zaptel Channel)
exten = s,n,Set(CHAN=${CHANNEL:4})
exten = s,n,Set(CHAN=${CUT(CHAN,-,1)})
exten = s,n,Macro(from-zaptel-${CHAN},${DID},1)
; If nothing there, then treat it as a DID
exten = s,n,NoOp(Returned from Macro from-zaptel-${CHAN})
exten = s,n,Goto(ext-did,${DID},1)
--
Thanks Regards,
Vidura B. Senadeera.
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Re: [asterisk-users] Sip Phone CID
try actually setting the rpid in the dialplan using sipcalledrpid(name,number) Rob Schall wrote: I set both the trustrpid and sendrpid to yes, but the calling phone still doesn't show the caller id of the person they are calling. Jason Fuermann wrote: check out rpid Mark Johnson wrote: Rob Schall wrote: This might sound like an odd question but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But... our old phone system set the caller id on the senders phone to show who they called. Example... If Sally calls Jim, then Sally's phone should just say 1001, it should say Jim 1001. Any know if this is possible. Our old PBX did this, and the bosses were curious if this is possible. Thanks, Rob I have tried over and over to figure out how to do this and it doesn't seem possible at the moment. I know this can be done with chan_sccp and maybe even chan_skinny (haven't tried that in a few years), but you'd need Cisco phones to do it. Is this something on anyone's To-Do list? Thanks, Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to limit IAX calls
Aaah... I'll try that...
A simple variable would be so much easier... :-)
BarZ
Jonathan k. Creasy wrote:
A demonstration:
exten = _X.,1,Set(GROUP()=${CALLERID(num))
exten = _X.,n,Set(CDR(AccountCode)=${CALLERID(num))
exten = _X.,n,GotoIf($[${GROUP_COUNT(${CALLERID(num))} 2]?103)
exten = _X.,n,Macro(trunk,${EXTEN},residential)
exten = _X.,n,Hangup
exten = _X.,103,Playback(allison7/all-circuits-busy-now)
exten = _X.,n,Hangup
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta
Sent: Friday, January 19, 2007 6:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to limit IAX calls
Take a look on:
Dialplan applications:
GetGroupMatchCount([EMAIL PROTECTED])
SetGroup([EMAIL PROTECTED])
Using this two applications you can deploy a max calls control inside
your dialplan!
check this too: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup
Hope it helps
On 1/19/07, Barzilai Spinak [EMAIL PROTECTED] wrote:
The SIP channels have a call-limit parameter (which is badly
documented and I haven't tested yet)
How can I have the same behaviour for IAX channels? I can't see anything
related to it.
Ah, I'm using Asterisk 1.2.13... maybe there is something in the 1.4
versions... but I can't change to 1.4 right now because of MFC/R2
BarZ
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[asterisk-users] Set Parameter of Call Files
Hi all,
I'm implementing call files and everything works nicely except that the
variable that I set in the call file does not seem to get populated.
Channel:SIP/MyProvider/910555
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context:myCallFileContext
Extension: s
Priority: 1
Set:myVar=MyNewValue
and...
[myCallFileContext]
exten=s,1,NoOp(${myVar}) ; == myVar is empty
I thought it might be related to write space since call files didn't
seem to work until inserted a tab (9) between param name and value. I
tried just using a single space as well, but it didn't seem to make a
difference.
I'm running 1.2.14 on CentOS4.4...
Is there something obvious that I have overlooked?
Thank you,
--
Warm Regards,
Lee
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Re: [asterisk-users] 1 phone 2 voicemail accounts
Hi all, I have a similar issue. I am looking for a way to make 1 phone to subscribe to 2 voicemail accounts on 2 different Asterisk machines on the same LAN linked over IAX2. Management requested that User1 on Asterisk1 should be able to forward a voicemail message to User2 on Asterisk2. All of our users are using GXP2000 SIP phones and the Asterisk servers communicate over IAX2. I will appreciate any help. Thank you and warm regards, Buki On 1/18/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: -- Forwarded message -- From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 18 Jan 2007 15:36:47 +0100 Subject: Re: [asterisk-users] 1 phone 2 voicemail accounts [EMAIL PROTECTED] wrote: What is the best way to have 1 phone check multiple voicemail accounts. I am using polycom 650 phones, and am wondering if mwi can work when checking multiple accounts. Try [EMAIL PROTECTED],[EMAIL PROTECTED] ; Subscribe to status of multiple mailboxes in sip.conf Best regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay in Call Distribution using the Queue Application
Yes, I confirm the autofill option is present in 1.4, but must be enabled manually not to break compatibility with 1.2. l. On Fri, 19 Jan 2007 15:32:32 +0100, Tom Rymes [EMAIL PROTECTED] wrote: You may be running into the limitation in Asterisk 1.2 (It's fixed in 1.4, I think double check that) in how the queues distribute calls. Basically, the queue can only distribute one call at a time, so if you have two agents, both available, and two calls in the queue, asterisk will send call #1 to agent #1 first. Once that call is connected, Asterisk will then send call #2 to agent #2. In other words, until asterisk distributes the first call, it can't distribute any other calls waiting in line. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
Bernardo Vieira wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Gordon Henderson wrote: If you only have one * box behind the NAT gateway then I don't really see a big issue with it to be honest. Port-forward on the firewall/router device (5060 and 1 through 2) to the * device, and use STUN on the client device to help it get through it's local NAT firewall/router. I use the same strategy and it works just fine, however, I did have an issue, here's the scenario: SIP Ext 1 +---+ |--- NAT -- Internet -- NAT -- Asterisk SIP Ext 2 +---+ Each of the SIP extensions work individually, but if I try to use both of them, only the first one registers. Bernardo, Just a thought: Try using a different SIP port for one of the extensions, if possible. Bob... ___ --Bandwidth 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] Red: Sip Phone CID
Here is what I have in my extensions.conf file now. Trustrcid and
sendrcid are turned to yes in the conf file.
I'm not fully sure how the SIPCalledRPID works though. The example I
found seems to try and provide the stuff automatically (id and name),
but does the SIPPEER stuff even exist?
I think this is probably the right track though. Any insight would be
much appreciated.
[macro-stdexten2];
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - Dial flags
exten =
s,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
exten = s,2,Dial(${ARG2},13,${ARG3}) ; Ring the interface, 20
seconds max
exten = s,3,Goto(s-${DIALSTATUS},1) ; Jump based on status
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail
exten = s-NOANSWER,2,Goto(incoming,s,1) ; If they press #, return to
start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail
(busy)
exten = s-BUSY,2,Goto(incoming,s,1) ; If they press #, return to
start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
answer
exten = a,1,VoicemailMain(${ARG1}); If they press *, send to
Voicemail
Thanks
Rob
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RE: [asterisk-users] Parked calls and the # key
Has there been a fix released for this? We have just upgraded to 1.4, and this issue still exists, and is a real pain. Thanks in advance. Gregory P. Scasny Golden Technologies, Inc. http://www.golden-tech.com blocked::http://www.golden-tech.com [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] 219-462-7200 - Ph. 574-233-1300 - Ph. (866) 806-7127 - Toll Free 219-462-7257 - Fax. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Thursday, January 11, 2007 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Parked calls and the # key I am perplexed by this so I how someone can help me out. On one of my servers the users began complaining that if they picked up a parked call they could not use the # key to transfer the call. This is a particualarly annoying issue since everyone has been taught to use #700 to park calls. At first I thought it was a DTMF issue with the polycom phones, since rebooting seemed to fix the problem. On further examination though, the # to transfer works on any call unless it has been retreievd from the parking lot. Then I began doubting that it had ever been possible, until I confirmed it worked on a system I built a few months back and someone in the IRC confirmed it was working on their systems. As it stands now I have taken configs from working and non working systems and compared them and have not found anything. I can reproduce this by calling someone, via sip phone or on a zap channel have the call parked by dialing #700 pickup the call, by dialing 701 or so. At this point the only transfer option I have is the transfer key on the phone, which works, but not as quick for the users. I'm using the standard parking, I simply included parkedcalls into a features context and all phones have access to it. show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer One Touch Monitor Disconnect Call * * pbx1*CLI Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-720 This is an svn checkout of 1.2 r49922 Can someone tell me what might cause this? -- Bruce 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] 1 phone 2 voicemail accounts
You cannot forward voicemails between Asterisk servers. Naija Man wrote: Hi all, I have a similar issue. I am looking for a way to make 1 phone to subscribe to 2 voicemail accounts on 2 different Asterisk machines on the same LAN linked over IAX2. Management requested that User1 on Asterisk1 should be able to forward a voicemail message to User2 on Asterisk2. All of our users are using GXP2000 SIP phones and the Asterisk servers communicate over IAX2. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT solutions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Bernardo, Just a thought: Try using a different SIP port for one of the extensions, if possible. Bob... Bob, Tanks for the tip. I had actually done that before, as a matte of fact that's the solution I have in place now. The thing is, even though it works it is not exactly the best solution as it forces some non-standard configuration on the clients. It's not my case, but imagine the hassled it would impose in an ITSP environment. Bernardo -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFsPHM2QVs8jsa1mQRArzQAJ4lZCEVhUX7zi8+rIQ79/335SeOzACdFPU4 nMXe9lmVnzLZKK6uTJqUlzo= =9N1q -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1 phone 2 voicemail accounts
A little php and SCP would make this work. You could do a web interface like vmail.cgi. -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Friday, January 19, 2007 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1 phone 2 voicemail accounts You cannot forward voicemails between Asterisk servers. Naija Man wrote: Hi all, I have a similar issue. I am looking for a way to make 1 phone to subscribe to 2 voicemail accounts on 2 different Asterisk machines on the same LAN linked over IAX2. Management requested that User1 on Asterisk1 should be able to forward a voicemail message to User2 on Asterisk2. All of our users are using GXP2000 SIP phones and the Asterisk servers communicate over IAX2. ___ --Bandwidth 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] Integrating asterisk with Toshiba Astrata DK380
Hi,
your zapata.con has 'context=from-pstn'
Try changing this to 'context=from-zaptel'
Robert Jenkins.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: 19 January 2007 15:19
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject: [asterisk-users] Integrating asterisk with Toshiba Astrata DK380
Deat all,
I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.
Following is my setup
Asterisk - Digium TE110P - E1 card in toshiba pbx - Toshiba PBX
A = B
C D
Asterisk PBX and strata PBX connected using back to back E1 cross cable.
Physicall connectivity is OK. The digium TE110p
LED state green. zttool also OK.
Toshiba stata configured to make outbound call via E1 link with pressing 9
and then the out side number.
I was able to make call from soft phone to analog extension at toshiba pbx.
A== B way as shown above. But when trying to dial from
Toshiba PBX analog extension to asterisk extension, by pressing 9 the call
rejected.
In the asterisk command prompt I'm having following error message.
-- Extension '' in context 'from-pstn' from '' does not exist. Rejecting
call on channel 0/31, span 1
Is there any wrong in my setup. dial plan??, additional configuration if i
required to put please guide me.
I will be greately appreciated your feedback on this regard.
configuration details
/etc/zaptel.conf
# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
/etc/asterisk/zapata.conf
signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=euroisdn
;switchtype=national
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number is
in milliseconds
callerid=asreceived
overlapdial=no
pridialplan=unknown
immediate=no
;rxwink=300
callprogress=no
loadzone=au
context=from-pstn ; Points to the default context of your extensions.conf
group=2
channel=1-15
channel=17-31 ;PRI/E1 link
[trunkgroups]
trunkgroup=2,16
spanmap=1,2,1
/etc/asterisk/extension.conf
[from-zaptel]
exten = _X.,1,Set(DID=${EXTEN})
exten = _X.,n,Goto(s,1)
exten = s,1,NoOp(Entering from-zaptel with DID == ${DID})
; If ($did == ) { $did = s; }
exten = s,n,Set(DID=${IF($[${DID}= ]?s:${DID})})
exten = s,n,NoOp(DID is now ${DID})
exten = s,n,GotoIf($[${CHANNEL:0:3}=Zap]?zapok:notzap)
exten = s,n(notzap),Goto(ext-did,${DID},1)
; If there's no ext-did,s,1, that means there's not a no did/no cid route.
Hangup.
exten = s,n,Macro(hangup)
exten = s,n(zapok),NoOp(Is a Zaptel Channel)
exten = s,n,Set(CHAN=${CHANNEL:4})
exten = s,n,Set(CHAN=${CUT(CHAN,-,1)})
exten = s,n,Macro(from-zaptel-${CHAN},${DID},1)
; If nothing there, then treat it as a DID
exten = s,n,NoOp(Returned from Macro from-zaptel-${CHAN})
exten = s,n,Goto(ext-did,${DID},1)
--
Thanks Regards,
Vidura B. Senadeera.
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[asterisk-users] Disconnect Supervision UK / BT solution?
Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : TDM400P amp; Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting hangups? Using BT? If so, contact BT and ask what the Disconnect Clear Time setting is for your phone line. Odds are it's probably 100. Increasing it to 800 fixed the issue for me. Disconnect Clear Time is BT's name for CPC. Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Comments appreciated before I get on the phone with BT -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Open Source Hosted PBX
Does anyone know if there exists an Open Source Hosted PBX platform based on asterisk? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue and Interface time out
Does anyone have ringinuse=no and autopause=yes working together in queues.conf? We assign members to our customer service queue from an application based on actions the agents take on their PCs. No static agents are defined in agents.conf and no members are specified in queues.conf. All member interfaces are SIP with only the basics configured in sip.conf. Even with 'ringinuse=no' configured, the Queue application continues to send callers to busy members causing them to get paused when their SIP device returns that it's busy. Does the Queue application need hints for member interfaces to determine their status? Thanks, James James Fromm wrote: No, call-limit is not being used. Do you have ringinuse=no working? Has anyone seen it work? Each SIP device has a very minimal config in sip.conf. Here's a show sip peer: * Name : 3207 Secret : Set MD5Secret: Not set Context : outbound Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : Sam 3207 MaxCallBR: 384 kbps Expire : 40 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 216.239.128.189 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 3207 SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (ulaw:20) Auto-Framing: No Status : OK (14 ms) Useragent: PolycomSoundPointIP-SPIP_650-UA/2.0.3.0131 Reg. Contact : sip:[EMAIL PROTECTED] Watkins, Bradley wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Thursday, January 18, 2007 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out I guess I'm missing something else. 'ringinuse = no' doesn't change anything. While on a call, the queue still sends another call and proceeds to set the member paused after receiving 'Busy Here' back from the SIP device. My queues.conf is: [general] persistentmembers = no [customerservice] persistentmembers = no musiconhold = default reportholdtime = no strategy = leastrecent timeout = 20 retry = 5 wrapuptime = 30 ;allow agents 30 seconds to wrap up work maxlen = 0 ;unlimited callers on hold servicelevel = 60 ;calls must be answered within 60 seconds announce-holdtime = no autopause = yes ringinuse = no joinempty = yes leavewhenempty = no Am I missing something obvious? What do your SIP peers look like? Are you using the call-limit feature? - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Red: Sip Phone CID
I'm not sure about the sippeer stuff, or where they get that variable.
We lookup our info in a database to set it. Also to use sipcalledrpid
you'll probably need the patch at http://bugs2.digium.com/view.php?id=6643 .
Rob Schall wrote:
Here is what I have in my extensions.conf file now. Trustrcid and
sendrcid are turned to yes in the conf file.
I'm not fully sure how the SIPCalledRPID works though. The example I
found seems to try and provide the stuff automatically (id and name),
but does the SIPPEER stuff even exist?
I think this is probably the right track though. Any insight would be
much appreciated.
[macro-stdexten2];
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - Dial flags
exten =
s,1,SIPCalledRPID(${SIPPEER(${EXTEN}:callerid_name)},${SIPPEER(${EXTEN}:callerid_num)})
exten = s,2,Dial(${ARG2},13,${ARG3}) ; Ring the interface, 20
seconds max
exten = s,3,Goto(s-${DIALSTATUS},1) ; Jump based on status
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail
exten = s-NOANSWER,2,Goto(incoming,s,1) ; If they press #, return to
start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail
(busy)
exten = s-BUSY,2,Goto(incoming,s,1) ; If they press #, return to
start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no
answer
exten = a,1,VoicemailMain(${ARG1}); If they press *, send to
Voicemail
Thanks
Rob
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Re: [asterisk-users] NAT solutions
Bernardo Vieira wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Bernardo, Just a thought: Try using a different SIP port for one of the extensions, if possible. Bob... Bob, Tanks for the tip. I had actually done that before, as a matte of fact that's the solution I have in place now. The thing is, even though it works it is not exactly the best solution as it forces some non-standard configuration on the clients. It's not my case, but imagine the hassled it would impose in an ITSP environment. Bernardo Bernardo, Yes. I have a home system running on Trixbox with an ITSP that provides two channels, but I can only register one of them as I cannot register twice on port 5060 with one IP address at each end. Both DIDs work, I just cannot determine which one a call is coming in on. The ITSP always reports the DID I registered last. The other drawback is the inability off-load the media path. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked calls and the # key
A fix was posted to the svn version of 1.2 and 1.4. The actual revision number I do not recall. On 1/19/07, Greg Scasny [EMAIL PROTECTED] wrote: Has there been a fix released for this? We have just upgraded to 1.4, and this issue still exists, and is a real pain. Thanks in advance. Gregory P. Scasny Golden Technologies, Inc. http://www.golden-tech.com [EMAIL PROTECTED] 219-462-7200 - Ph. 574-233-1300 - Ph. (866) 806-7127 snap://%28866%29%20806-7127 - Toll Free 219-462-7257 - Fax. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Bruce Reeves *Sent:* Thursday, January 11, 2007 2:29 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Parked calls and the # key I am perplexed by this so I how someone can help me out. On one of my servers the users began complaining that if they picked up a parked call they could not use the # key to transfer the call. This is a particualarly annoying issue since everyone has been taught to use #700 to park calls. At first I thought it was a DTMF issue with the polycom phones, since rebooting seemed to fix the problem. On further examination though, the # to transfer works on any call unless it has been retreievd from the parking lot. Then I began doubting that it had ever been possible, until I confirmed it worked on a system I built a few months back and someone in the IRC confirmed it was working on their systems. As it stands now I have taken configs from working and non working systems and compared them and have not found anything. I can reproduce this by calling someone, via sip phone or on a zap channel have the call parked by dialing #700 pickup the call, by dialing 701 or so. At this point the only transfer option I have is the transfer key on the phone, which works, but not as quick for the users. I'm using the standard parking, I simply included parkedcalls into a features context and all phones have access to it. show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer One Touch Monitor Disconnect Call * * pbx1*CLI Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-720 This is an svn checkout of 1.2 r49922 Can someone tell me what might cause this? -- Bruce 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 -- Bruce 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
[asterisk-users] Re: Background + Dial
Hmm...I have just noticed this issue as well I want the background command to play soundfiles while the dialplan moves on and is dialing a number of zap channels etc... It plays, but essentially ends up being no different than Playback() I note now before posting this, that the Background is intended to ALLOW caller input (as in an IVR) while sound is playing I guess I am confused as to the Background command's purpose ... Any ideas for what I want to do ? -- Chris [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Thanks for your reply, But I want to have an interactive menu, not just a music. So, the customer can have information menu while he's waiting the call is answer. I'dont now if it's possible with MoH. Thanks a lot -- Initial Header --- From : [EMAIL PROTECTED] To : 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Cc : Date : Tue, 27 Jun 2006 18:58:26 +0700 Subject : RE: [Asterisk-Users] Background + Dial Hi GL Pls. config MOH and use Dial command with m option. This will allow you execute Dial command while providing Music in the background. Hope it help Hoa Thai Duy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, June 27, 2006 5:01 PM To: asterisk-users Subject: [Asterisk-Users] Background + Dial Hi everybody, I try this : [incoming_from_fxo_card] exten = s,1,Answer() exten = s,2,Background(filename) exten = s,3,Dial($(INTERNAL_SIP_TEL)) But * wait the file is finish before make Dial to SIP channel. Background(filename) (from voip-info.org) = Starts playing a given sound file, but immediately returns, permitting the sound file to play in the background while the next commands (if any) execute. I want to Dial a SIP channel while playing sound and waiting for a digit from a ZAP channel. In other words, i want to make a interactive MoH while waiting for the SIP channel answer. Is it possible? Thanks a lot and excuse me for my poor english (I'll fix this in few months). GL - ALICE SECURITE ENFANTS - Protégez vos enfants des dangers d'Internet en installant Sécurité Enfants, le contrôle parental d'Alice. http://www.aliceadsl.fr/securitepc/default_copa.asp ___ --Bandwidth 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 - ALICE SECURITE ENFANTS - Protégez vos enfants des dangers d'Internet en installant Sécurité Enfants, le contrôle parental d'Alice. http://www.aliceadsl.fr/securitepc/default_copa.asp ___ --Bandwidth 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] mISDN
Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. Tnx Giordano -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.0/639 - Release Date: 18/01/2007 18.47 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: One way choppy sound
On 2007-01-17 10:29:43 -0800, Yelson Vivas [EMAIL PROTECTED] said: Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)==sip==(asterisk 1)iax2 trunk(asterisk 2) ===alaw==(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i don't understand why the sound is bad in only one way. Any sugestions to solve it more than welcome Usually sounds can be choppy one way due to constrained upstream bandwidth. There might be plenty of room for the audio to get to you, but that doesn't mean the reverse is at all true. Jitter buffering can help this, or using a more compact format (like GSM or g729) is also a potential helper. Good luck, hope this helps, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked calls and the # key
Bruce Reeves wrote: A fix was posted to the svn version of 1.2 and 1.4. The actual revision number I do not recall. Bruce, Have you had a chance to test? I've been looking for the last hour for the bug report I thought you posted earlier this week, but haven't found anything. 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] mISDN
On Fri, 2007-01-19 at 18:51 +0100, Giordano Grandis wrote: Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. # yum install bc Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect Supervision UK / BT solution?
Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Just to be clear, what is the exact disconnect problem that you see? I have three TDM cards in two different systems, one using PBX lines and one on a private BT line. Both of them have trouble detecting a caller who is ringing, but then hangs up before being answered by the asterisk system However, *all* of them are absolutely fine at spotting a normal hangup once the call is connected and I see no random disconnects during calls either. Can you confirm that this is what you mean, or whether it's something else? Ed W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked calls and the # key
Here is the bug: http://bugs.digium.com/view.php?id=8804 I have not tried the svn version yet, I modified my features.c file as noted in the bug and have been running it successfully. On 1/19/07, Doug Lytle [EMAIL PROTECTED] wrote: Bruce Reeves wrote: A fix was posted to the svn version of 1.2 and 1.4. The actual revision number I do not recall. Bruce, Have you had a chance to test? I've been looking for the last hour for the bug report I thought you posted earlier this week, but haven't found anything. 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 -- Bruce 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] mISDN
Friday, January 19, 2007, 7:06:40 PM, Patrick wrote: On Fri, 2007-01-19 at 18:51 +0100, Giordano Grandis wrote: Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. # yum install bc if # yum install bc -/bin/bash: yum: command not found try apt-get install bc -- Best regards, Csibra Gergomailto:[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] Parked calls and the # key
Bruce Reeves wrote: Here is the bug: http://bugs.digium.com/view.php?id=8804 Thank you very much! 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] Parked calls and the # key
Yesthanks a billion. I am trying your patch on our 1.4 install right now Thanks again Greg Gregory P. Scasny Golden Technologies, Inc. http://www.golden-tech.com [EMAIL PROTECTED] 219-462-7200 - Ph. 574-233-1300 - Ph. (866) 806-7127 - Toll Free 219-462-7257 - Fax. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Friday, January 19, 2007 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parked calls and the # key Bruce Reeves wrote: Here is the bug: http://bugs.digium.com/view.php?id=8804 Thank you very much! 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 ___ --Bandwidth 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] MIT Using Asterisk - VM Server
I saw this and laughed, the toilet server, AKA the voice mail server is running Asterisk. http://www.networkworld.com/news/2007/011907-mit-your-take.html?page=1 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Parked calls and the # key
Worked like a champ Thanks again... Gregory P. Scasny Golden Technologies, Inc. http://www.golden-tech.com blocked::http://www.golden-tech.com [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] 219-462-7200 - Ph. 574-233-1300 - Ph. (866) 806-7127 - Toll Free 219-462-7257 - Fax. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Friday, January 19, 2007 12:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parked calls and the # key Here is the bug: http://bugs.digium.com/view.php?id=8804 I have not tried the svn version yet, I modified my features.c file as noted in the bug and have been running it successfully. On 1/19/07, Doug Lytle [EMAIL PROTECTED] wrote: Bruce Reeves wrote: A fix was posted to the svn version of 1.2 and 1.4. The actual revision number I do not recall. Bruce, Have you had a chance to test? I've been looking for the last hour for the bug report I thought you posted earlier this week, but haven't found anything. 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 -- Bruce 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] Parked calls and the # key
Bruce Reeves wrote: Here is the bug: http://bugs.digium.com/view.php?id=8804 I have not tried the svn version yet, I modified my features.c file as noted in the bug and have been running it successfully. Looks like I'll have to manually make the changes as well. The current 1.2 Branch seems to have broken transfers 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] Queue and Interface time out
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Friday, January 19, 2007 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out Does anyone have ringinuse=no and autopause=yes working together in queues.conf? We assign members to our customer service queue from an application based on actions the agents take on their PCs. No static agents are defined in agents.conf and no members are specified in queues.conf. All member interfaces are SIP with only the basics configured in sip.conf. Even with 'ringinuse=no' configured, the Queue application continues to send callers to busy members causing them to get paused when their SIP device returns that it's busy. Does the Queue application need hints for member interfaces to determine their status? Thanks, James Queue does not need hints, but it does need the channel driver (in your case SIP) to inform it whether or not the member interface is in use. That is actually why I asked about call-limit. Can you try adding a call-limit (even if it's 10 or 20 or whatever) and see if that solves your problem? Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how can PRI, BRI and analog cards achieve a synchronous clock / timing
hello list, i have a problem regarding the synchronisity (clock source) when using multiple cards. e.g. when having connected one PRI port of our TE410P to the telco, i need to have the analog card like the TDM400P or a B410P synchronous to the clock of our telco provider. otherwise faxing on the analog cards does not work or i get cracking noise or even hangups on my BRI lines, due to bit slips. as long as the ports are on the same pci-card, they're synchronous, but not when one has to use another card (e.g. having a PRI telco line and some analog fax machines or some BRI ISDN equipment served by asterisk) junghanns or beronet have a solution for this (PCM port on the card; they can even switch the voicedata over this bus), but i can't find any solution for digium cards. i've found a timing connector on the TE410P. can this somehow be utilized? is there another (software) solution? am i the only one with this problem (haven't found anything about this on the mailing list) thanks frank sautter ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] how can PRI, BRI and analog cards achieve a synchronous clock / timing
If you use a channel bank like the Adtran Atlas 550, you can specify a primary sync to the telco, and every subsequent connection to the Atlas uses that sync as a timing source. Expensive, but I expect you can pick one up or something like it on Ebay. Nothin beats an Atlas, though. -Original Message- From: Frank Sautter [mailto:[EMAIL PROTECTED] Sent: Friday, January 19, 2007 12:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how can PRI, BRI and analog cards achieve a synchronous clock / timing hello list, i have a problem regarding the synchronisity (clock source) when using multiple cards. e.g. when having connected one PRI port of our TE410P to the telco, i need to have the analog card like the TDM400P or a B410P synchronous to the clock of our telco provider. otherwise faxing on the analog cards does not work or i get cracking noise or even hangups on my BRI lines, due to bit slips. as long as the ports are on the same pci-card, they're synchronous, but not when one has to use another card (e.g. having a PRI telco line and some analog fax machines or some BRI ISDN equipment served by asterisk) junghanns or beronet have a solution for this (PCM port on the card; they can even switch the voicedata over this bus), but i can't find any solution for digium cards. i've found a timing connector on the TE410P. can this somehow be utilized? is there another (software) solution? am i the only one with this problem (haven't found anything about this on the mailing list) thanks frank sautter ___ --Bandwidth 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] Ebay Unwired Buyer, Using Asterisk?
Just got a call from Ebay's unwired buyer and The Voice is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real competition). 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
Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?
Allison is not exclusively working for asterisk, she also does other recordings. Zao Steve Totaro wrote: Just got a call from Ebay's unwired buyer and The Voice is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real competition). 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?
If you think that constitutes a competitive edge then there isn't much help. Personally I think it's great that everyone can leverage off a 'shared' resource like Allison. At least you know it will sound professional and be recorded at a high standard. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 19 January 2007 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ebay Unwired Buyer, Using Asterisk? Just got a call from Ebay's unwired buyer and The Voice is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real competition). 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 ___ --Bandwidth 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] Set(X=10|g) vs Set(GLOBAL(X)=10)
Hi, show application Set says: ---cut--- Set(name1=value1|name2=value2|..[|options]) [...] g - Set variable globally instead of on the channel ---cut--- But someone told me that the proper way is to use Set(GLOBAL(X)=10). Is Set(X=10|g) somewhat deprecated or not? I know that both run fine on 1.4 but I'd like to know which one is better to be sure to be ready for future upgrades. Stefan -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 19 Jan 2007, at 21:07, Steve Totaro wrote: Just got a call from Ebay's unwired buyer and The Voice is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real competition). There was a link posted to an interview with Allison a few weeks back. She mentioned eBay as a customer, and how she used eBay unwired before and and listened to herself speak. It doesn't mean they use Asterisk, though. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFFsSoDRAx5nvEhZLIRAojzAJwKfbZGsuFQO45ds0+ZY0jh4wYhawCfT5q1 MT40p83x78dm0CIxQGpNh8c= =4aFS -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?
Dean Collins wrote: help. Personally I think it's great that everyone can leverage off a 'shared' resource like Allison. Now that's a nice wording. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk - http://www.das-asterisk-buch.de ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] J1/INS1500 and the Redirect Number
and what if you do:
zap debug span 2
OR:
zap intense debug span 2
Is the value anywhere there?
On 1/19/07, Gary Mensenares [EMAIL PROTECTED] wrote:
Hello Andrew!
Thanks for taking the time to reply.
Sorry but no, it doesnt seem to show up.
Here's my dial plan:
exten = _X.,s,Answer()
exten = _X.,n,Set(Ani=${CALLERID(ani)})
exten = _X.,n,Set(Dnis=${CALLERID(dnis)})
exten = _X.,n,Set(Rdnis=${CALLERID(rdnis)})
exten = _X.,n,NoOp(ANI: ${Ani})
exten = _X.,n,NoOp(DNIS: ${dnis})
exten = _X.,n,NoOp(RDNIS: ${rdnis})
exten = _X.,n,SayDigits(${Ani})
exten = _X., Hangup()
This is what I get:
[Jan 19 14:20:14] DEBUG[24089]: chan_zap.c:7649 pri_find_empty_chan: Found
empty available channel 0/23
-- Accepting call from '090' to '4590' on channel 0/23, span 2
[Jan 19 14:20:14] DEBUG[24089]: chan_zap.c:1458 zt_enable_ec: Enabled echo
cancellation on channel 47
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/47-1, ) in new stack
[Jan 19 14:20:14] DEBUG[24083]: channel.c:895 channel_find_locked: Avoiding
initial deadlock for channel '0x8b86720'
[Jan 19 14:20:14] DEBUG[7387]: pbx.c:1621 pbx_substitute_variables_helper_full:
Function result is '090'
-- Executing [EMAIL PROTECTED]:2] Set(Zap/47-1, Ani=090) in new
stack
[Jan 19 14:20:14] ERROR[7387]: func_callerid.c:91 callerid_read: Unknown
callerid data type 'dnis'.
[Jan 19 14:20:14] DEBUG[7387]: pbx.c:1621 pbx_substitute_variables_helper_full:
Function result is ''
-- Executing [EMAIL PROTECTED]:3] Set(Zap/47-1, Dnis=) in new stack
[Jan 19 14:20:14] DEBUG[7387]: pbx.c:1621 pbx_substitute_variables_helper_full:
Function result is ''
-- Executing [EMAIL PROTECTED]:4] Set(Zap/47-1, Rdnis=) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(Zap/47-1, ANI: 090)
in new stack
-- Executing [EMAIL PROTECTED]:6] NoOp(Zap/47-1, DNIS: ) in new stack
-- Executing [EMAIL PROTECTED]:7] NoOp(Zap/47-1, RDNIS: ) in new
stack
-- Executing [EMAIL PROTECTED]:8] SayDigits(Zap/47-1, 090) in
new stack
However, we are quite positive that the redirect number is in there because on
mobile terminals, both the redirect and the ani are identified.
Anyone out there had any luck with the RDNIS before?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Joakimsen
Sent: Friday, January 19, 2007 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] J1/INS1500 and the Redirect Number
Is it not coming in as CallerID(RDIS)? The specifications for
the service don't seem too different from any other PRI.
On 1/16/07, Gary Mensenares [EMAIL PROTECTED] wrote:
Hi everyone!
I'm wondering if anyone on the list had the opportunity to
work with
an NTT INS1500 ISDN PRI service before.
You see, in Japan, if you receive a call that was just forwarded by
another number, the call presentation not only includes the caller
(ANI) and your number (DNIS), it will also usually include the
forwarding number (REDIRECT). Does anybody know how to
extract this field on Asterisk?
For reference, you can look at
http://www.ntt-east.co.jp/ISDN/tech/spec/espec/3-5/content_3.html.
This is the full specification of INS 1500 signalling. Any
assistance
would be very much appreciated.
Thanks again!
Sincerely,
Jug Mensenares
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[asterisk-users] Re: Audiocodes GPL
Ok, so why was the GPL license violated? Or did Mr. Torvalds offer you an alternate license arrangement? On 1/17/07, Evan Kirstel [EMAIL PROTECTED] wrote: Sorry for the confusion..the MP202 is running Linux! -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 16, 2007 4:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; # Legal; Yaniv Nizan; Evan Kirstel Subject: Audiocodes GPL I have some Audiocodes units which appear to be running Linux, according to the unit's own System Log kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 However my contact at Audiocodes claims otherwise On 12/4/06, Yaniv Nizan [EMAIL PROTECTED] wrote: I doubt that we are running Linux on the MP-202. Perhaps there is a reference to the OS on the PC that configures the device So a few questions: 1) Does anyone know if the older Audiocodes devices (such as the multiport gateways) run Linux as well? 2) What does one go about doing to correct GPL violations? Perhaps someone has a generic legal letter that can be used in these situations? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN
Ciao Giordano, Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. You have to install the bc package, according to your distribution's package manager. HTH, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] using the Manager to connect caller to conference
Is there a way, via the Asterisk Manager, to dial an extension and connect it with an existing meetme conference? I'm trying to pull callers into a conference as other conference members leave. Thanks in advance. -George ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Red: Sip Phone CID
Jason Fuermann wrote: I'm not sure about the sippeer stuff, or where they get that variable. We lookup our info in a database to set it. Also to use sipcalledrpid you'll probably need the patch at http://bugs2.digium.com/view.php?id=6643 . I looked at this in the past and never made it work correctly. Does this work in the newest version of 1.2? Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Audiocodes GPL
On 12/4/06, Yaniv Nizan [EMAIL PROTECTED] wrote: 2) What does one go about doing to correct GPL violations? Perhaps someone has a generic legal letter that can be used in these situations? This should help answer that question: http://www.fsf.org/licensing/licenses/gpl-faq.html#ReportingViolation -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power users ~ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set Parameter of Call Files
Lee Jenkins wrote: Hi all, I'm implementing call files and everything works nicely except that the variable that I set in the call file does not seem to get populated. Channel:SIP/MyProvider/910555 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: myCallFileContext Extension: s Priority: 1 Set: myVar=MyNewValue and... Resolved. Tried again using tabs between value/pairs and it works. Not sure what I did wrong originally though... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided 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 Manager and Ruby
(Sorry for the way-late response to this short thread...) We use rami in production on an Asterisk 1.2.3 server, and have had basically zero problems at least since 1.2.3 was released. rami and ruby's built in RPC provide a very easy to use proxy, if you have multiple clients which all need access to the AMI as we do. But yeah, I'd expect Asterisk has diverged a lot since rami was last updated. I did a round of refactoring at the time we were initially developing our screen pop app, but none of it has had to change in over a year. Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] On Wed, 1 Nov 2006, snacktime wrote: On 11/1/06, Rajkumar S [EMAIL PROTECTED] wrote: Hi, Any one using Rubi asterisk manager interface http://rubyforge.org/projects/rami/ ? How stable/usable it is? It probably hasn't seen much use. I created that back when I was just learning ruby, so it could probably use some refactoring as well. And If anything has changed in the asterisk manager protocol that would be an issue also. I created it against the beta version at the time, can't remember what that was. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: Polycom buddies question
Hints in extensions.conf in conjuction with mac-directory.xml with bw set to 1. Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Doug Sent: Thu 1/18/2007 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Polycom buddies question At 11:56 1/18/2007, Bill Gibbs wrote: Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C73B2A.03C9AD84 A follow up (late better than never) Finally had time to sit down and look at this sip.cfg keys key.scrolling.timeout=1 key.IP_500.31.function.prim=BuddyStatus/ This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! How do you know which buddy is being monitored? Does this show a screen of buddies? Bill -- From: Bill Gibbs Sent: Thursday, December 07, 2006 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: Polycom buddies question Figures I email this and realized I can hit Menu 1 (Features) 4 (Presence) 2 (Buddy Status) Wow that's a lot of key strokes. Anyway to reduce that to a one button touch? I don't mind doing that but I guess I should think of the users J Bill -- From: Bill Gibbs Sent: Thursday, December 07, 2006 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Polycom buddies question I know this is not asterisk specific but we all know this group is used for Polycom issues as well. I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this. Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: Polycom buddies question
yes it shows the normal Buddies screen that is available from the LCD if that feature is enabled in the Polycom sip config file (presence) Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Doug Sent: Thu 1/18/2007 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Polycom buddies question At 11:56 1/18/2007, Bill Gibbs wrote: Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C73B2A.03C9AD84 A follow up (late better than never) Finally had time to sit down and look at this sip.cfg keys key.scrolling.timeout=1 key.IP_500.31.function.prim=BuddyStatus/ This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! How do you know which buddy is being monitored? Does this show a screen of buddies? Bill -- From: Bill Gibbs Sent: Thursday, December 07, 2006 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: Polycom buddies question Figures I email this and realized I can hit Menu 1 (Features) 4 (Presence) 2 (Buddy Status) Wow that's a lot of key strokes. Anyway to reduce that to a one button touch? I don't mind doing that but I guess I should think of the users J Bill -- From: Bill Gibbs Sent: Thursday, December 07, 2006 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Polycom buddies question I know this is not asterisk specific but we all know this group is used for Polycom issues as well. I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this. Bill ___ --Bandwidth 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] Red: Sip Phone CID
Yes it should, I'm not running bleeding edge 1.2 but it isn't an older branch either. Mark Johnson wrote: Jason Fuermann wrote: I'm not sure about the sippeer stuff, or where they get that variable. We lookup our info in a database to set it. Also to use sipcalledrpid you'll probably need the patch at http://bugs2.digium.com/view.php?id=6643 . I looked at this in the past and never made it work correctly. Does this work in the newest version of 1.2? Mark ___ --Bandwidth 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] Incoming SIP line does not display CallerID correctly
Hi all,
I've just setup a sip line with Telasip and when they route the calls to
my asterisk box, they include an extension along with the context that
is defined in sip.conf for that DID.
At first, I couldn't figure why they were getting 404 error from my
asterisk box, but then figured out that they are sending the call to an
extension that matches my number with them, in the context defined in
sip.conf. So instead of transferring the call to say, incoming they
are sending the call to incoming/55 where 55 is the
sip phone number I have with them.
So, I have to account for that extension with something like this:
[incoming]
exten=55,Goto(incoming,s,1)
Thus transferring the call to the context that I want it to come in on.
The problem that I have is the caller ID ${CALLERID(num)} always shows
the actual number provided by Telasip and not the actual caller id
information.
I also have axVoice and they do not do it this way. They simply send it
to the context without specifying an extension.
Below is a sip packet. The Caller ID comes through correctly on the sip
packet by for some reason as I mentioned, Asterisk is reporting it as
the number I have with the sip provider.
Below is the sip packet. The 33 represents my cell phone I
was using to call into the system, which was correct.
localhost*CLI exit
-- SIP read from 4.79.19.56:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Record-Route: sip:4.79.19.56;ftag=as5bfea671;lr=on
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=0
Via: SIP/2.0/UDP
4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK3f2e414e;rport=5060
From: 33 sip:[EMAIL PROTECTED];tag=as5bfea671
To: sip:[EMAIL PROTECTED];tag=as12b47a8d
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Telasip GW3
Max-Forwards: 69
Remote-Party-ID: 33
sip:[EMAIL PROTECTED];privacy=off;screen=no
Content-Length: 0
P-hint: proxy loose route
--
Warm Regards,
Lee
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[asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 79
Hi,
I checked by changing to from-zaptel, but no luck yet. Pls guide me on
this.
Regards,
vudura senadeera
--
Message: 9
Date: Fri, 19 Jan 2007 16:47:18 -
From: Robert Jenkins [EMAIL PROTECTED]
Subject: RE: [asterisk-users] Integrating asterisk with Toshiba
Astrata DK380
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Hi,
your zapata.con has 'context=from-pstn'
Try changing this to 'context=from-zaptel'
Robert Jenkins.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: 19 January 2007 15:19
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject: [asterisk-users] Integrating asterisk with Toshiba Astrata
DK380
Deat all,
I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.
Following is my setup
Asterisk - Digium TE110P - E1 card in toshiba pbx - Toshiba PBX
A = B
C D
Asterisk PBX and strata PBX connected using back to back E1 cross cable.
Physicall connectivity is OK. The digium TE110p
LED state green. zttool also OK.
Toshiba stata configured to make outbound call via E1 link with pressing
9
and then the out side number.
I was able to make call from soft phone to analog extension at toshiba
pbx.
A== B way as shown above. But when trying to dial from
Toshiba PBX analog extension to asterisk extension, by pressing 9 the
call
rejected.
In the asterisk command prompt I'm having following error message.
-- Extension '' in context 'from-pstn' from '' does not
exist. Rejecting
call on channel 0/31, span 1
Is there any wrong in my setup. dial plan??, additional configuration if
i
required to put please guide me.
I will be greately appreciated your feedback on this regard.
configuration details
/etc/zaptel.conf
# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
/etc/asterisk/zapata.conf
signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=euroisdn
;switchtype=national
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your
needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number
is
in milliseconds
callerid=asreceived
overlapdial=no
pridialplan=unknown
immediate=no
;rxwink=300
callprogress=no
loadzone=au
context=from-pstn ; Points to the default context of your
extensions.conf
group=2
channel=1-15
channel=17-31 ;PRI/E1 link
[trunkgroups]
trunkgroup=2,16
spanmap=1,2,1
/etc/asterisk/extension.conf
[from-zaptel]
exten = _X.,1,Set(DID=${EXTEN})
exten = _X.,n,Goto(s,1)
exten = s,1,NoOp(Entering from-zaptel with DID == ${DID})
; If ($did == ) { $did = s; }
exten = s,n,Set(DID=${IF($[${DID}= ]?s:${DID})})
exten = s,n,NoOp(DID is now ${DID})
exten = s,n,GotoIf($[${CHANNEL:0:3}=Zap]?zapok:notzap)
exten = s,n(notzap),Goto(ext-did,${DID},1)
; If there's no ext-did,s,1, that means there's not a no did/no cid
route.
Hangup.
exten = s,n,Macro(hangup)
exten = s,n(zapok),NoOp(Is a Zaptel Channel)
exten = s,n,Set(CHAN=${CHANNEL:4})
exten = s,n,Set(CHAN=${CUT(CHAN,-,1)})
exten = s,n,Macro(from-zaptel-${CHAN},${DID},1)
; If nothing there, then treat it as a DID
exten = s,n,NoOp(Returned from Macro from-zaptel-${CHAN})
exten = s,n,Goto(ext-did,${DID},1)
--
Thanks Regards,
Vidura B. Senadeera.
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Message: 10
Date: Fri, 19 Jan 2007 11:46:57 -0500
From: Chris Earle \(CBL\) [EMAIL PROTECTED]
Subject: [asterisk-users] Disconnect Supervision UK / BT solution?
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Hi all
I'm using sangoma a200 cards in the UK and have the ongoing, often noted
problem of disconnect supervision with BT POTS lines.
Just noticed this post on
http://www.voip-info.org/wiki/view/UK+Asterisk+Details
stating that potentially someone's got a solution :
TDM400P amp; Not Detecting Hangups:
Got a TDM400P installed and having problems with Asterisk not detecting
hangups? Using BT? If so, contact BT and ask what the Disconnect Clear
Time setting is for your phone line. Odds are it's probably 100.
Increasing
it to 800 fixed the issue for me.
Disconnect Clear Time is BT's name for CPC.
Does anyone have any thoughts/confirmation about this finally being a
viable
solution? This disconnect supervision problem has plagued TDM and
Sangoma
cards for a long time!
Comments appreciated before I get on the phone with BT
Re: [asterisk-users] Queue and Interface time out
That worked. I don't understand what call-limit has to do with this. I set it to 5. Why does that keep the member interface from getting a second call from the Queue application? I would think it would allow the member interface to get up to 5 calls. Watkins, Bradley wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Friday, January 19, 2007 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out Does anyone have ringinuse=no and autopause=yes working together in queues.conf? We assign members to our customer service queue from an application based on actions the agents take on their PCs. No static agents are defined in agents.conf and no members are specified in queues.conf. All member interfaces are SIP with only the basics configured in sip.conf. Even with 'ringinuse=no' configured, the Queue application continues to send callers to busy members causing them to get paused when their SIP device returns that it's busy. Does the Queue application need hints for member interfaces to determine their status? Thanks, James Queue does not need hints, but it does need the channel driver (in your case SIP) to inform it whether or not the member interface is in use. That is actually why I asked about call-limit. Can you try adding a call-limit (even if it's 10 or 20 or whatever) and see if that solves your problem? Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided 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 the Manager to connect caller to conference
see Originate manager Action in voip-info.org On 1/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Is there a way, via the Asterisk Manager, to dial an extension and connect it with an existing meetme conference? I'm trying to pull callers into a conference as other conference members leave. Thanks in advance. -George ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone know what this warning is about? Nothing in list history about it either..
On inbound calls from my SIP provider I get multiple warnings as follows: WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid host Everything else works but these warnings are a pain and I don't know what they are about Nothing on previos lists or Google explains... ___ --Bandwidth 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 1.4 and g723
I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan applications I have tested except for Echo. The critical application for us is Voicemail. When a call to voicemail extension is initiated the Asterisk console does not indicate any error. Packet captures indicate the call is active and streaming g723 data. Everything seems well but is not. The audio at the client is unrecognizable. One thing that I have noticed is that the bitrates in the upstream and downstream direction differ. From Asterisk to ATA the g723 bitrate switches between 5.3 kb/s and 6.3 kb/s. From ATA to Asterisk the bitrate is a constant 6.3 kb/s. I don't think this is a problem but seems odd. As a comparison I captured packets from a call to the echo application and found that the bitrate was 6.3 kb/s in both upstream and downstream packets. Additionally, all prompts are g723 format. Voicemail is saved as g723sf. As a parrallel task a co-worker has gotten 1.2 to work with g723. However we require 1.4 for t.38 pass-through. The end-to-end system is illustrated below. [Asterisk] / \ ip ip / \ [PSTN]--pri--[GATEWAY]--ip--[ATA]--2pr--[Phone] System details -Asterisk server version 1.4 - compiled from source - Fedora Core 6 -Gateway - Cisco 2811 -ATA - Linksys 2102 I would appreciate any advice or suggestions. It should be noted that the calls to the PSTN through the gateway and calls between ATA's are working fine. Regards, Phil French Phil French Systems Engineer --- CapRock Communications 4400 S. Sam Houston Parkway E. Houston, Texas 77048 Office: 832 668 2643 [EMAIL PROTECTED] www.caprock.com NOTICE OF CONFIDENTIALITY: This e-mail message may contain confidential information and is intended only for the person(s) named above. Any review, use, disclosure or distribution by any other person is prohibited. If you are not the intended recipient, please contact the sender by e-mail and destroy all copies of this message. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chanskype
Hi, I tried the try version of chanskype, however, everytime that I make a call asterisk generate an error Anyone has experience with this? Since I tried to contact the support but they never replied. 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] CAS on Sangoma boards
Dear folks, I would be very thankful if an experienced user can help me out here. I wanna use mfcr2 and unicall library on sangoma boards but so far impossible for me. As Im setting the framing type to CAS (TE_SIGMODE = CAS) on wanpipe I couldnt get the link alarm out (i looped a A102d links) but when setting it to CCS everything works fine and the green lights shine on the back. Can anyone send me a working sample of wanpipe.conf and zaptel.conf for cas signalling? and is it possible that the alarms are because of looping the links (although in ccs mode it works just fine) ? Any help and hint would be highly appreciated. PS. i define the span in zaptel as cas with hdb3 -- M. Shokuie Nia. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
