Re: [asterisk-users] USB Cordless
Hi Jeremy, Sorry, I have no clue about your question, but I have a question in regards to your USB Cordless handsets. Do you have any idea on what you will be using ? We've tested a couple of thems so far, but I'm still searching better products. So if you have any ideas about any Wireless handsets I'd love to know which products you're planning to use. Thanks! Jeremy Mann a écrit : Does anyone know if X-Ten or SJPhone support multiple cordless handsets for multiple lines? I have an office with multiple roaming users(nurses) that are in and out. I'd like to provide them telephones, and my idea is to have a PC sitting in a corner somewhere running a softphone client. When a nurse comes in she just picks up any available handset(anywhere from 2-5 per office) and starts calling. Each handset would be labeled with their extension so that if any inbound calls came to them they'd be able to let the receptionist know their extension. Any ideas? Also, is it possible to transfer a call directly to someone's VM(if they are out of the office) bypassing their extension? If so, could someone post the asterisk logic behind the extension setup? I don't want anything too complex(like setting the DND or phone to busy). Thanks. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by *MailScanner* http://www.mailscanner.info/, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceXML + Nuance
Well, basically, I'm looking for something that has the possiblity to use the Nuance licenses, and that can do Text to Speech, as well as Voice Recognition. So far it doesn't seem possible to have a single product that does all this within Asterisk... Rob Townley a écrit : Voxy - the only way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done. Is something the free and open source Voxy what you are looking for? http://sourceforge.net/projects/voxy On 5/3/07, *wendell hamilton* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I've done considerable work with the voxeo Prophecy platform, and it's been successful, albeit challenging at times. -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Eric Rousse Sent: Thursday, May 03, 2007 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VoiceXML + Nuance Hello, Is there anyone who has ever done a setup of VoiceXML combined with some licenses from Nuance for the ASR/TTS engine within Asterisk ? I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS engine, but we are having a couple of issues which I guess are caused by VoiceGenie. If there's an alternative, it would be very interesting for us. Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com http://www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceXML + Nuance
Hello, Is there anyone who has ever done a setup of VoiceXML combined with some licenses from Nuance for the ASR/TTS engine within Asterisk ? I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS engine, but we are having a couple of issues which I guess are caused by VoiceGenie. If there's an alternative, it would be very interesting for us. Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - VoiceGenie IVR
Hi,
I'm currently working on a setup between Asterisk and VoiceGenie (which
is a IVR system).
The way my setup is done, is that I have a PRI line coming in my
Asterisk server, and then VoiceGenie is connected to Asterisk via SIP,
like any other softphone basically. I'm able to receive calls in
Asterisk and then link them with VoiceGenie. But one of my issues is
that when I get an outside call, transfer the call to VoiceGenie, then
for that specific calls VoiceGenie would decide that this call has to be
transfered to an outside party, so then VoiceGenie calls up that number,
it goes through Asterisk and it reached the other person. But the link
doesn't stay up very long, max 15 seconds.
That's one of the errors that I see in Asterisk(for obvious reasons I've
replaced some numbers with *):
-- Hungup 'Zap/8-1'
Feb 15 14:10:19 WARNING[25664]: chan_sip.c:1227 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 1
(Critical Response)
Feb 15 14:10:28 WARNING[25664]: chan_sip.c:1227 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 1
(Critical Response)
-- Hungup 'Zap/1-1'
Here's a part of my dialplan for outside calls:
exten = _9XX,1,Set(CALLERID(all)=450-655-)
exten = _9XX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
And here's a Macro that I use for incoming call for VoiceGenie:
[macro-voicegenie]
exten = s,1,Answer
exten = s,2,SIPAddHeader(X-Asterisk-DID: ${ARG1})
exten = s,3,SIPAddHeader(X-Asterisk-CallerName: ${ARG2})
exten = s,4,Dial(SIP/108)
exten = 514380,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514380,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514373,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514373,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
exten = 514373,1,Macro(voicegenie,${EXTEN},${CALLERID(name)})
Here's the config in sip.conf:
[108]
type=friend
context=internal
host=10.1.1.40
callerid=VoiceGenie 108
progressinband=never
disallow=all
allow=ulaw
Also, the support team at Voicegenie they asked me if I stop sending
183 Session Progress before 180 Ringing.
It seems that this could be part of my issue.
Thanks,
--
Eric Rousse
System Administrator
514.380.2992
450.655.1001
1.888.641.5800
Telmatik inc.
204 Montarville, suite 250
Boucherville, QC, Canada
J4B 6S2
www.telmatik.com
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[asterisk-users] Dell Servers
Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Servers
Hello, Well we're planning either use some PRI lines or IP Trunk, where not sure yet. For the PRI lines we will probably use a Wildcard TE412P, so PCIe For the IP Trunk, not sure yet I don't have a lot of info in that regards. I'm also planning to put an extra server with some cards to connect to a SAN with Fiber channel, not decided yet between Fiber channel and Gigabit switch dedicated. Anyway... Matt Florell a écrit : Hello, I have installed Asterisk on several of them and there can be issues. Will you be installing a telco interface card on this server?(If so, which one) Will this server have PCI or PCIexpress expansion ports? MATT--- On 2/1/07, Eric Rousse [EMAIL PROTECTED] wrote: Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the onboard controller and purchase an addon ethernet card. Does anyone has real experience ? Thanks, -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Rousse System Administrator 514.380.2992 450.655.1001 1.888.641.5800 Telmatik inc. 204 Montarville, suite 250 Boucherville, QC, Canada J4B 6S2 www.telmatik.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Storing recordings
Hello, I'm currently facing a decision regards to the system I have to build. Basically, I'm aiming for 2 Asterisk servers with 1 PRI line in each. And each of them will record all calls in and out. I was wondering if anyone had any suggestions in that regards ? I'm currently thinking of building these 2 servers, with some Dell PowerEdge 2950 and either connected directly to a Gigabit switch that would be connected to another machine that will have a lot of drive space. I could maybe mount some network drives on the 2950. Or maybe I could connect the 2950 to the other machine with a Fiber cable ? But i'm still undecided if I should use a SAN or NAS. I guess that in overall and if I wanna build something though enough to stay there for a least 3-4 years, I should go for a SAN. Anyone any thoughts ? First time I'm building a such system, I have good knowledge about it since I already used some in the past, but if anyone has experience with Asterisk and such system it would be great to share your experience! Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Toll free numbers
Hi,
For some reason, I seem to have issues with dailing toll free numbers
and can't seem to find out why, sometimes, I get a busy signal. Some
other times I get weird errors from the phone.
The error below was a simple busy signal.
Here's couple of my info relevant to the problem:
-- Reconfigured channel 1, PRI Signalling signalling
-- Reconfigured channel 2, PRI Signalling signalling
-- Executing Dial(SIP/107-9da02970, Zap/g1/18889554562) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/18889554562
-- Zap/1-1 is proceeding passing it to SIP/107-9da02970
-- PROGRESS with cause code 28 received
-- Zap/1-1 is making progress passing it to SIP/107-9da02970
-- Hungup 'Zap/1-1'
== Spawn extension (internal, 918889554562, 1) exited non-zero on
'SIP/107-9da02970'
from the console I get this error Progress with cause code 28 received.
In my extensions.conf file I got this in my internal context which is
used by my sip phones:
[internal]
include = trunktollfree
include = outgoing
exten = _XXX,1,Macro(incoming,SIP/${EXTEN},${EXTEN})
exten = 200,1,Answer
exten = 200,2,MusicOnHold()
exten = 999,1,Playback(demo-echotest)
exten = 999,2,Echo
exten = 999,3,Playback(demo-echodone)
exten = 1000,1, Dial(IAX2/1000,30)
;Agent login
exten = 3001,1,AgentCallbackLogin(||[EMAIL PROTECTED])
;Agent logout
exten = 3002,1,AgentCallbackLogin(||l)
exten= 2020,1,Answer
exten= 2020,2,Ringing
exten= 2020,3,Wait(2)
exten= 2020,4,Queue(queue1)
exten= 2020,5,Hangup
It's a bit messy but it's mainly for testing.
In trunktollfree, I got this:
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten = _91800.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
zaptel.conf:
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone=us
And I'm using a PRI line on my server, outgoing calls are working good,
it's just my toll free calls that doesn't go through, I've probably
misconfigured something I guess...
Thanks!
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[asterisk-users] Unique ID
Hello guys, We're currently working on asterisk trying to create our own SIP phone, because we need special features. But dunno maybe there's other people who already done that before. Basically, we are a inbound call center. We have serveral customers with different phone numbers, which are redirected to us. When we receive a call coming on a specific phone number, the call gets identified with the number and there's a greeting associated and displayed on the agent soft phone(this technology is still using regular phone with a special computer device). But here's the challenge that we currently face: 1. We need to have the info for the hold time (from agent) and hold time(before the call is actually answered). We currently offer a different pricing for the hold time by an agent than from the other one hold time. 2. We're currently trying to identify calls by unique id for billing, I've found about that the variable $UNIQUEID which I could use, and there's also the cdr table that I can create, but it would be nice to have both in the cdr table ? That way I could probably create a second table in the asterisk db, and store our hold time, sent from the softphone. Anyway, does all that ring a bell to someone ? Something that was already done ? Let me know if I'm unclear. Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] include context
Hello, Just wondering if there's a simple way to display the hierarchy of the includes within the extensions file ? Currently I have the sample file extension.conf in my Asterisk machine. But its kinda hard to search through the file to get the idea of the context hierarchy. Like which contexts call which ones. And which ones are not used. Is there a way of doing this ? Or is it just un-necessary for some reason ? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip show peers
Hello guys, Is there anyone who could explain me some stuff about sip show peers ? 108/10810.1.1.40 5060 OK (1 ms) 107/10710.1.1.246 D 51074OK (101 ms) The port seems different here, and the main difference is that the extension 108, is a server with a fixed IP 107, is a client with a softphone (X-Lite) and a dynamic IP. Why the diffrence in the port ? And why the difference in the reponse time ? We are on the same physical network, a ping is giving me a response of 1ms for each. Is it because the softphone is with a dynamic IP and Asterisk is treating this differently ? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip show peers
Hi Andrew, Thanks for the response. Interesting. But one thing though, both extensions are softphones actually. The one on 108, is actually VoiceGenie that I'm testing with Asterisk. But I'm trying to explain why I'm getting some glitch with the systems sometimes with my softphone, and I thought the response time of 101ms, was the answer. Is there anything I can do to improve that response time ? Thanks, Andrew Kirch a écrit : Response below -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Rousse Sent: Thursday, September 14, 2006 10:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip show peers Hello guys, Is there anyone who could explain me some stuff about sip show peers ? 108/10810.1.1.40 5060 OK (1 ms) 107/10710.1.1.246 D 51074OK (101 ms) The port seems different here, and the main difference is that the extension 108, is a server with a fixed IP 107, is a client with a softphone (X-Lite) and a dynamic IP. Why the diffrence in the port ? And why the difference in the reponse time ? We are on the same physical network, a ping is giving me a response of 1ms for each. Is it because the softphone is with a dynamic IP and Asterisk is treating this differently ? Thanks, SNIP A SIP ping and an ICMP ping are two different entities. The SIP ping operates at a higher level in the OSI stack than a simple ICMP ping. This means that whatever is receiving the ping has to do more work to decode it, and respond. I wouldn't worry about the latency difference as the SIP Ping is prioritized a bit more by a computer which is multitasking than by a hard phone which is not. As for the port, they simply chose to negotiate on a higher port. You might check your X-Lite settings, but I don't think this will break anything! Andrew ___ --Bandwidth 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] Queue Management
Hello, I'm working in a small call center, but with special requirements. We currently have a couple of clients, all of them have specific phone numbers configured in our system, so when we get a call for a specific client we take down the information via a webpage then it sent via email to them. One of the major problem that I'm seeing is the queue management. Right now with our current system, the agents are able to see what call are coming in, which one haven't been answered, which one are on hold. (That part is not so bad with Asterisk since it's already taking care of this) But the part I'm worring about is that the agent can see the Greeting message for the customer line. So the agent knows what to say before answering the line then IE popups with the URL for that client. Not sure if that can be replicated with Asterisk. We could probably adapt our selfs by doing a query about the DNIS and then store the DNIS associated with his greeting. Almost like what we do now actually... The only thing is to put all that together... hehehe ___ --Bandwidth 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] High Availability
Hey guys, I'm currently investigating solutions about High Availability solution, I've found out about this webpage on voip-info.org: http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions But that's cool for the voice and stuff. But what about the recording. If I don't want to put too much load on the main servers that will process the calls, is there a way that we can setup 2 extra servers, that will handle the call recording ? I'm guessing that call recording might generate a lot of work and i/o on the disk. But not sure, since I've never done that before and not even sure if its a good idea to transmit that over the network ? Anyway, if anyone has any other link about High Availability Solutions, that would be great! Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple issues
Hello guys,
Not sure if it's me or what, but I'm starting to learn Asterisk. And I'm
currently reading the Oreilly book and another one. And I was at the
point to test the s extension. But when I try to use it it doesn't work
and the call gets rejected in Asterisk.
Here's a part of my extension.conf file:
[default]
exten = 5143802603,1,Dial(SIP/444)
include = incoming
[internal]
include = outgoing
exten = 1204,1,Dial(SIP/1204)
exten = 400,1,Dial(SIP/400)
exten = 401,1,Dial(SIP/401)
exten = 402,1,Dial(SIP/402)
exten = 403,1,Dial(SIP/403)
exten = 404,1,Dial(SIP/404)
exten = 500,1,Dial(Zap/8/5147815376)
exten = 801,1,Answer()
[incoming]
exten = s,1,Answer()
exten = s,2,Background(enter-ext-of-person)
exten = 1204,1,Playback(digits/1204)
exten = 1204,2,Dial(SIP/1204,10)
exten = 1204,3,Playback(vm-nobodyavail)
exten = 1204,4,Hangup()
[outgoing]
exten = _XX,1,Dial(${OUTBOUNDTRUNK})
N.B.: A couple of extension are only for testing.
And here's some specific part about zapata.conf
context=default
switchtype=5ess
signalling=pri_cpe
First, the incoming context, since its included in the default context,
and I've configured the zapata.conf file, to use the default context for
the PRI line,
I thought that when I call was coming in to the PRI line, it was going
to go through the default context, if the case wasn't found there it
would read the include file.
And it doesn't seem to reach the s extension.
But if I remove the s extension and configure the phone number that are
on my lines, it works. That's weird, not sure if I've done something
incorrectly somewhere.
Second problem, If I change the s extension for a specific number the
first steps works, but after that I get a hang up on me. I've replicated
an exemple from the
book and it doesn't seem to work, maybe because of some specific config
with the line ?
Third problem, my outgoing context doesn't seem to work, I always get
like a fast busy signal.
Thanks for any help!
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[Asterisk-Users] Asterisk and TBCT
Hi Guys, I'm starting to work on Asterisk, trying to see if it will fit our needs, but so far it seems it doesn't support TBCT(Two B Channel Transfer). I've found a couple of links that was talking about TBCT, and someone had posted a bounty for that feature, but no news since 2003. I've also seen that it seems there is some support in libpri, or work was done around 2005 for TBCT and there's a mention about supporting TBCT but only with 5ESS. And at the moment I don't have the equipement to try it out, so before we do try it out I was trying to investigate a bit. In the case that there's no support for TBCT, is there other possible way we can do it ? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
