Re: [asterisk-users] Asterisk core dumps files
On Mon, Jul 27, 2009 at 01:28:59PM -0300, Gustavo A Gonzalez wrote: Hello all! Im running asterisk 1.4.23 and sometimes it crashes. Because I need to look for what asterisk crashes I run asterisk with option '-g' for debugging purpose. When I search for core files in filesystem nothing happend and I have not generated core files. Which is the way to know if asterisk are generating core dump files? And Which is the directory where it saves them? Is necessary to recompile asterisk with some extra option? Thanks for any idea. IIRC it tries to dump them in the directory it was run from. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP vs Analog lines
First off, you should post a new message rather than replying to an existing message. On Tue, Jul 28, 2009 at 08:18:42PM -0400, John F. Ervin wrote: Never having actually rolled an Asterisk (Trixbox in my case) system into production. I was wondering if in most peoples opinion if given the choice would rather have a straight VOIP/SIP system or would rather have a system with normal POTS/analog types lines and something like a digium card? As far as reliability etc. Thoughts? Just to mention the third option: a digital PSTN line. Not as ugly as an analog PSTN line, but generally more expensive. If you're in the US, this is probably only meaningful if you need at least c. 8 outgoing lines (maximal number of concurrent outgoing calls) but I figure this generally varies wildly. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP client Resp code
hello,
I have SIP phone registered with my server
now if they send me any number for dialing then i want to give a response
code
actually this number is conference number and i need to chek via DB query
that this conference is valid or not
if conference is not valid then i want to send a response code to SIP device
that conf not valid
i tried with Sendtext but not getting success
here are my dialplan
exten = _X.,1,NOOP(${CALLERID(num)}${CALLERID(DNID)}--)
exten = _X.,n,SendText(Hello You are Now connected)
exten = _X.,n,SipAddheader(RESP_CODE : Hello You are Now connected)
exten = _X.,n,Meetme(151515,sdMAwC)
exten = _X.,n,Hangup()
can anybody help this out
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Re: [asterisk-users] SIP vs Analog lines
On Tue, 28 Jul 2009, John F. Ervin wrote: Never having actually rolled an Asterisk (Trixbox in my case) system into production. I was wondering if in most peoples opinion if given the choice would rather have a straight VOIP/SIP system or would rather have a system with normal POTS/analog types lines and something like a digium card? As far as reliability etc. Thoughts? A lot may well depend on where you are in the world, how reliable the local PSTN is vs. the local broadband - or leased line service. There will be a cost trade-off too - VoIP over broadband generally being cheaper than installing multiple POTS/ISDN2/ISDN30 connections, but for a slightly lower guarantee of service. Here in the UK we have a generally excellent wholesale (ADSL) broadband system, let-down at times by the back-end ISPs who buy into it, so picking a good back-end ISP is worth it. And how many channels or concurrent calls are you looking at? So too many variables to give a definate answer, but personally I'm installing more and more pure VoIP systems these days. Even when the client wants/needs to keep their legacy PSTN, we use the PSTN for inbound and VoIP for outbound calls. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Misunderstood thing
So, there's two kinds of authentication that routinely go on in the SIP client/server world: 1) REGISTER authentication -- this is the 401 Unauthorized challenge to an initial REGISTER request that causes it to be resent with WWW-Authorize headers containing various authentication credentials, including an encrypted nonce. ClientServer - REGISTER #1 -- -- 401 Unauthorized --- REGISTER #2 + creds -- 200 OK -- This authentication is for *registration* *only*. It does *not* authorise you to place outbound calls. It only provides a mechanism for authenticating your request to secure a contact binding to a certain AOR (address of record) on the SIP server side, in order to receive calls at that AOR. Allowing calls to be placed on the basis that the originator is merely registered is not sufficiently secure. I could place calls as you by spoofing your username (AOR) as long as you are registered from somewhere else. 2) INVITE authentication -- the 407 Proxy Authorization Required challenge. The mechanism is very similar in its anatomy to the registration challenge, but is applied to an INVITE you originate toward the server instead. ClientServer -- INVITE #1 --- 407 proxy challenge - INVITE #2 + creds -- 100 Trying - - other prov. msgs etc -- -- 200 OK --- ... Anyway, this deals with registration authentication (scenario #1): [general] register = user:p...@server:5060 And this deals with INVITE request authentication (scenario #2): username=user secret=pass Does that help? -- Alex -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Matching Originate action with its NewChannel event
An application commanding asterisk with AMI is going to launch lots of concurrent calls in very few seconds using the Originate AMI command but it's also going to need to be able to cancel very quickly any call of them even before each OriginateResponse event comes in. All the calls will be done by the same trunk (a trunking enabled channel). But there's a problem for canceling any call: there's no way to know what channel to hangup to because all channel prefixes in the NewChannel event are the same (the trunking channel one) and although the Originate action has an ActionId property, it isn't available in the NewChannel event but only in the OriginateResponse event, being very late. I've read some of you are using the CallerId property but in this case it's not an option because the application needs to establish the same callerId for all of them. Is there any solution using AMI? I'm planning to use asterisk 1.4.18 Thanks Jose ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail attachments not working
When you removed the mailboxes, you either messed up permissions or just
made the files unavailable to Asterisk. You presumably still have
/var/spool/asterisk/voicemail/default, so you need to check that tree vs the
tree on a working server.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Rojas
Sent: Tuesday, July 28, 2009 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail attachments not working
Hello,
Your smtp server is on?
Best regards
Carlos Rojas
On Mon, Jul 6, 2009 at 7:30 PM, Steve Anness steve.ann...@gmail.com wrote:
Today I discovered that voicemail attachments are not working on our latest
asterisk server (version 1.4.24.1). I have two other asterisk servers that
I maintain but I didn't do the configuration on these so this is my first
time that I have done the voicemail.conf. I get an email but there is no
attachment. Maybe there is something else I need to configure that I don't
know about? Here is my actual config, the only difference is I removed all
the mailboxes for the purpose of sharing with the world. However, I have
made sure there are not spaces between fields as I hear that causes
problems.
[general]
format = gsm|wav49|wav
attach = yes
serveremail = asterisk
serveremail = nore...@mustangintl.com
mailcmd = /usr/sbin/sendmail -v -t -f aster...@hisg-it.net
maxlogins = 3
emaildateformat = %A, %B %d, %Y at %r
sendvoicemail = yes ; Allow the user to compose and send a voicemail while
inside
emailsubject = [PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
[zonemessages]
eastern = America/New_York|'vm-received' Q 'digits/at' IMp
central = America/Chicago|'vm-received' Q 'digits/at' IMp
central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european = Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
[default]
116 = 1149,employee,emplo...@domain.org
Suggestions?
Thank you everyone in advance.
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Re: [asterisk-users] Possibly I don't understand sip peers
On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote: I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a 404. shouldn't I be able to put in a kind of wildcard for his IP block or am I just being silly? If not, what am I doing wrong? I think you've got your syntax wrong there... permit and deny statements are used to create Access Control Lists and to limit the IP address ranges. The allow and disallow statements are to allow or disallow various codecs. They way you've specified it above, you're allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably isn't what you want. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Instant messaging (yeah, again)
Hi, over the year, the IM question popped up here and then, the answer was allways something along the line yeah, there's an experimental patch, but it's not yet ready... Are there any plans to change this, to ie. allow messages between xmpp and sip or iax? Especially with the chan_skype (that's hopefully is going to be released in my lifetime ;)) a messaging infrastructure in the core would be interesting... Regards, Jay ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HPEC VPM ?
Hi - I had a client recently move their asterisk system (asterisk 1.4.26, dahdi 2.2.0.1, aex800 w/vpm module) to a new location, a building that's nearly 150 years old. I was not personally able to go there, but the person who did the move said the building's demarc room was scary-- water leaks, jumbled and frayed wiring, and all sorts of other fun. The echo on their POTS lines has proven to be quite problematic. The hybrids are balanced, txgain and rxgain are optimized individually for all channels, and the vpm module on the card is doing its job. For many calls, this has been effective. Still, echo remains on calls to some destinations, particularly those on the closer exchanges. On calls to one particular number, if I turn the echo canceler off, the echo sounds as loud and clear as if the destination was actually echo(). With the echo canceler on, echo is still very pronounced. The echo tail is clearly longer than 16 ms. I even tried disabling the vpm module (vpmsupport=0 in base.c) and using oslec instead with a setting of echocancel=512. After a long convergence period, oslec seemed to do a slightly better job than the vpm module, but echo was still bad enough to make a conversation nearly impossible. My question for anyone with knowledge on this: would HPEC do a better job than the VPM module (or oslec)? Can HPEC cope with very long echo tails? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP vs Analog lines
Steve Totaro escribió: On Tue, Jul 28, 2009 at 9:13 PM, Miguel Molina mmol...@millenium.com.co mailto:mmol...@millenium.com.co wrote: John F. Ervin escribió: Never having actually rolled an Asterisk (Trixbox in my case) system into production. I was wondering if in most peoples opinion if given the choice would rather have a straight VOIP/SIP system or would rather have a system with normal POTS/analog types lines and something like a digium card? As far as reliability etc. Thoughts? I'd go VoIP without thinking twice. Always think twice and always look both ways before crossing the street. Look left, right, and then left again Very true, I just wanted to emphasize that VoIP with asterisk is a great alternative on many scenarios. We are on the 21st century! Many technological efforts that have been made through all this years have been directed to bring telephony to the IP world. While true, I also have read that unless major upgrades are done, P2P, YouTube, other streaming, and tons of other bandwidth intensive apps are going to bog down the net in many spots. Hopefully it is not one of your hops to your ITSP. That's why QoS exists. Make P2P bog down, not golden VoIP packets. Again it's about network design, management and quality of the provider. IHMO running VoIP on the open Internet is possible, but doing it carefully, not throwing just hundreds of simultaneous calls to see how well they work. And accepting that not always the quality will be the best. If you can't accept the downs of it, ask for a dedicated link between your places. Asterisk has played and keeps playing a pretty nice role on the open source market we are in. VoIP will be as reliable and good quality as your network is. Your network, your ISPs, or your provider? If it is just your network then you must be speaking of TDM. I'm speaking of LAN part of the network doing VoIP calls, where the quality of switches and good design are key, and where the bandwidth is definitely plenty and free. The savings of not having to make double phone/data cabling and the advantages of VoIP are now a standard worldwide, from carriers to small home PBXs. Most new cable jobs run cat5 or cat6 regardless of use for almost the same price. I actually don't know of any cabling outfits offering cat3. Most existing workspaces have data jacks already in place. Analog lines are definitely legacy. The last time I put a T1 channel bank into use was more than two years ago, and never had to configure another one since then. I think he is just referring to a small amount of lines, although he did not say explicitly. I don't know about a channel bank (except for a whole bunch of fax machines) Yeah that was a little less than the full 24 analog lines connected to the channel bank. That got replaced with some VoIP phones and softphones connected to the asterisk PBX of the company. Finally I was comparing analog POTS lines to VoIP in PBX applications, where the differences are huge in terms of configuration and infrastructure efforts, features, and with use of telephony cards, reliability. TDM is another story, and better than VoIP on some aspects, like stable audio quality, good detail of hangup causes (Q.931) compared to SIP response codes and so on. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possibly I don't understand sip peers
Jared Smith wrote: On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote: I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a 404. shouldn't I be able to put in a kind of wildcard for his IP block or am I just being silly? If not, what am I doing wrong? I think you've got your syntax wrong there... permit and deny statements are used to create Access Control Lists and to limit the IP address ranges. The allow and disallow statements are to allow or disallow various codecs. They way you've specified it above, you're allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably isn't what you want. Your looking for host=dynamic. Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ignoring time spent waiting in queue in CDR
Alex Balashov wrote: I wouldn't approach this by trying to rework the CDRs at all; CDRs are fundamentally low-level call records. They correspond to calls. If you need logic to support a billing model for some specific application (i.e. time after connect to agent), I would approach that from a higher layer of abstraction that is more closely coupled to the application's own. For example, you could listen for Manager API events that indicate a queue caller's connection to an agent and flag those. There are numerous ways to skin this cat. What I would not do is try to mess with the CDRs to achieve this end; there is a reason they are called CDRs -- call detail records. Not queue detail records, not MoH detail records, not IVR detail records, but _call_ detail records. If nothing else, you may find that someday you will need the total call duration for other purposes, and have shot yourself in the foot by hacking it out this way. Plus, it's just too hard. Why jerry-rig CDRs when there are far easier and more functionally modular / extended ways to accomplish the same goal? Wrong tool for the job. Just my $.02, of course... Scott Gifford wrote: Hello, I'm working on an Asterisk configuration for a call center, and they bill based on the time spent talking to an agent, but not for any time spent waiting in a queue. The CDR information contains the entire duration of the call as billable seconds, including time spent waiting in the queue. I would like the billable seconds to only include the time spent actually talking to an agent. I am using Asterisk 1.4.18. The only way I have found so far is to correlate the CDRs with the CONNECT queue records, figure out the end time of the call by adding the CDR start time to the duration, then figure out the actual duration by subtracting the time of the queue CONNECT record. That seems messy and error-prone, and I'm hoping there's a better way. I also looked at using the ResetCDR() or ForkCDR() dialplan functions, but I don't see a way to cause code to run immediatly after the agent answers a call from the queue. Any suggestions? Am I missing some easy way of doing this? Thanks! Scott. ___ -- 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 I agree, I personally do this using the queue events from the AMI. Make sure you turn on queue events in queues.conf! Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk quick question
Thank you guys, for this clarification... Arturo Ochoa Electrosystems 2009/7/28 Kevin P. Fleming kpflem...@digium.com Miguel Molina wrote: Counting that everything works well on the IP portion of the communication, you might have something, but the store and forward process that has to be made twice to emulate a T.38 gateway on both asterisks would make it a very slow process to send a single fax, having that faxing is traditionally slow. So, it's pretty much a little more elaborate answer of unlikely to be pleasant. A direct T.38 gateway for asterisk would be awesome in any case of use. This is all correct. At this time, Asterisk 1.6.x (and thus Fax For Asterisk) don't offer a gateway mode, only sending and receiving of FAXes from TIFF files on the Asterisk system itself. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC VPM ?
Noah Miller wrote: My question for anyone with knowledge on this: would HPEC do a better job than the VPM module (or oslec)? Can HPEC cope with very long echo tails? HPEC and the Digium VPMADT032 use the same algorithms from the same vendor. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching Originate action with its NewChannel event
Jose Arias cyr2...@gmail.com writes:
An application commanding asterisk with AMI is going to launch lots of
concurrent calls in very few seconds using the Originate AMI command but it's
also going to need to be able to cancel very quickly any call of them even
before each OriginateResponse event comes in. All the calls will be done by
the
same trunk (a trunking enabled channel). But there's a problem for
canceling any call: there's no way to know what channel to hangup to because
all channel prefixes in the NewChannel event are the same (the trunking
channel
one) and although the Originate action has an ActionId property, it isn't
available in the NewChannel event but only in the OriginateResponse event,
being very late.
I had a similar problem with faxes, and at the suggestion of somebody
on the list, solved it like this: I sent a custom identifier variable
in with the AMI command, which was then available to the channel. I
directed the calls into a custom dialplan which used UserEvent to send
an event with the identifier variable and all of the channel
information. The script could then use this event to associate the
identifier it generated with the channel information for that call.
I just used the fax filename as a unique identifier, and I passed it
in with a Variable line to AMI, called FaxFile. The dialplan entry
was like this:
exten = s,1,UserEvent(FaxStarted|Channel:
${CHANNEL}|Uniqueid:${UNIQUEID}|FaxFile: ${FAXFILE})
I then matched everything up by lookin at the FaxFile part of the user
event. In your case, you could just make up a unique identifier of
your own to send.
Hope this helps,
Scott.
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Re: [asterisk-users] HPEC VPM ?
My question for anyone with knowledge on this: would HPEC do a better job than the VPM module (or oslec)? Can HPEC cope with very long echo tails? HPEC and the Digium VPMADT032 use the same algorithms from the same vendor. Aha. Thanks for this tidbit, Kevin! Next question: does anybody know how to handle extremely long tail echo that a VPM module cannot? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC VPM ?
Noah Miller wrote: Next question: does anybody know how to handle extremely long tail echo that a VPM module cannot? How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can handle 128ms echo tails, which is pretty darn long. It's rare to see an echo tail longer than that except on very high latency connections, or when the echo is actually acoustically generated by the far end and not by network effects. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording Calls
Greetings to all. this is my first question, and but that nothing is for consulting if this with asterisk can be realised. I have a commutator 3com, connected to 20 telephones of the same mark, my necessity right now is to be able to record the calls that enter the commutator, and wanted to know if this is possible with asterisk. by its attentions, thank you very much ___ Carlos Rodriguez Torreon Coahuila _ Messenger cumple 10 años y tiene regalos para ti www.aniversariomessenger.com.mx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording Calls
if you are running Asterisk in front of the other pbx you can record the calls that you send to the other system. You will either need some type of pri interface to connect between the two systems if digital and some fxs interfaces if analog. Tom _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mr. Rodriguez Sent: Wednesday, July 29, 2009 3:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Recording Calls Greetings to all. this is my first question, and but that nothing is for consulting if this with asterisk can be realised. I have a commutator 3com, connected to 20 telephones of the same mark, my necessity right now is to be able to record the calls that enter the commutator, and wanted to know if this is possible with asterisk. by its attentions, thank you very much ___ Carlos Rodriguez Torreon Coahuila _ Messenger cumple 10 años de ser parte de tu vida ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording Calls
Check OrecX out. The non GPL version may be able to do NBX protocol. Not sure what a 3com commutator unless that is what Digium and 3com teamed up on. Thanks, Steve On Wed, Jul 29, 2009 at 3:35 PM, Tom Moore tommym2...@gmail.com wrote: if you are running Asterisk in front of the other pbx you can record the calls that you send to the other system. You will either need some type of pri interface to connect between the two systems if digital and some fxs interfaces if analog. Tom -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mr. Rodriguez *Sent:* Wednesday, July 29, 2009 3:14 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Recording Calls Greetings to all. this is my first question, and but that nothing is for consulting if this with asterisk can be realised. I have a commutator 3com, connected to 20 telephones of the same mark, my necessity right now is to be able to record the calls that enter the commutator, and wanted to know if this is possible with asterisk. by its attentions, thank you very much ___ *Carlos Rodriguez *Torreon Coahuila -- Messenger cumple 10 años de ser parte de tu vidahttp://www.aniversariomessenger.com.mx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC VPM ?
Next question: does anybody know how to handle extremely long tail echo that a VPM module cannot? How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can handle 128ms echo tails, which is pretty darn long. It's rare to see an echo tail longer than that except on very high latency connections, or when the echo is actually acoustically generated by the far end and not by network effects. I haven't done any real measurement on it, but I believe it's actually longer than 128ms. As I go higher and higher with echocancel values, the echo does get better, but is never totally eliminated. At echocancel=1024, there is still rather pronounced echo on calls in the local exchanges. The calls are also more or less half-duplex at that point because the vpm is filtering out so much of the signal as echo. I may just tell the client to look at a partial PRI. All this echo chasing is getting costly for them. Thanks! Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recording calls again
They excuse my question again. I explain to them. I have an equipment 3com v3000 with 20 extensions, my question is if I can use asterisk to record the calls, is necessary an additional servant? at the moment only I have the telephones connected to this equipment. Where encounter information of how doing it? Carlos Rodriguez _ Gracias Messenger por estos 10 años www.aniversariomessenger.com.mx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording calls again
Mr. Rodriguez wrote: They excuse my question again. I explain to them. I have an equipment 3com v3000 with 20 extensions, my question is if I can use asterisk to record the calls, is necessary an additional servant? at the moment only I have the telephones connected to this equipment. Where encounter information of how doing it? Are you using Google Translate or Babelfish? -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording calls again
Some times i used babelfish :( Carlos Rodriguez Date: Wed, 29 Jul 2009 17:39:33 -0400 From: sean.bri...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Recording calls again Mr. Rodriguez wrote: They excuse my question again. I explain to them. I have an equipment 3com v3000 with 20 extensions, my question is if I can use asterisk to record the calls, is necessary an additional servant? at the moment only I have the telephones connected to this equipment. Where encounter information of how doing it? Are you using Google Translate or Babelfish? -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Con Windows Live, puedes organizar, editar y compartir tus fotos. http://www.microsoft.com/mexico/windows/windowslive/products/photo-gallery-edit.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Open Source Pavilion at AstriCon: Your project wanted!
Each year at AstriCon, we have an Open Source Pavilion which showcases projects which are adjuncts to Asterisk, or which are directly relevant to improving the utility and features of Asterisk. It gives smaller projects the chance to have some space to display what they're doing, and to give demonstrations to the AstriCon attendees who are interested in some of the things happening in the open-source ecosystem that surrounds Asterisk. I'd like to mention that we have a few open spots in the Open Source Pavilion still, and I'd like to solicit ideas for people who would like to have the opportunity to put your OSS project in front of the AstriCon attendee audience. What your project should have: - No significant corporate sponsorship - Open Source License - Running code - Integration or usefulness with Asterisk, relevant to attendees What you need for the booth: - Banner of some sort, with your project name/URL - Single-sheet handouts (both sides, preferably) with information on your project (200-300 sheets) - A knowledgeable person staffing the booth at any time that the show floor is open - Hopefully some sort of demo or hands-on display What you get: - A 6' table on the Expo floor - Electricity - Wireless Internet access - Two chairs - A huge crowd of highly intelligent, curious people who want to know what you're doing - A pass for 2 for the expo (or a special discount rate if you want to upgrade to sessions) We have a very limited number of tables, though we hope to get many projects on board. There is a chance your project may not be approved even though it's the coolest thing since sliced bread - but we'll try to get as many people on board as possible. Please don't take it personally if we run out of space before we run out of project proposals! There are so many great OSS things out there, but so few square feet. :-) Please email me directly with your project details to be included on the list of applicants. JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possibly I don't understand sip peers
Jared Smith wrote: On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote: I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a 404. shouldn't I be able to put in a kind of wildcard for his IP block or am I just being silly? If not, what am I doing wrong? I think you've got your syntax wrong there... permit and deny statements are used to create Access Control Lists and to limit the IP address ranges. The allow and disallow statements are to allow or disallow various codecs. They way you've specified it above, you're allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably isn't what you want. I have the codec permissions in the columns allow and disallow. Those seem to work ok. it's permit/deny/mask I seem to be having a problem with. Like I say, I don't think I understand their use or perhaps they don't work in realtime ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possibly I don't understand sip peers
Anthony wrote: Jared Smith wrote: On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote: I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a 404. shouldn't I be able to put in a kind of wildcard for his IP block or am I just being silly? If not, what am I doing wrong? I think you've got your syntax wrong there... permit and deny statements are used to create Access Control Lists and to limit the IP address ranges. The allow and disallow statements are to allow or disallow various codecs. They way you've specified it above, you're allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably isn't what you want. Your looking for host=dynamic. Anthony Tried that. dynamic seems to require a registration to work. Carriers don't register. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC VPM ?
Noah Miller wrote: Next question: does anybody know how to handle extremely long tail echo that a VPM module cannot? How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can handle 128ms echo tails, which is pretty darn long. It's rare to see an echo tail longer than that except on very high latency connections, or when the echo is actually acoustically generated by the far end and not by network effects. I haven't done any real measurement on it, but I believe it's actually longer than 128ms. As I go higher and higher with echocancel values, the echo does get better, but is never totally eliminated. At echocancel=1024, there is still rather pronounced echo on calls in the local exchanges. The calls are also more or less half-duplex at that point because the vpm is filtering out so much of the signal as echo. I may just tell the client to look at a partial PRI. All this echo chasing is getting costly for them. Thanks! Noah In cases of really short loops, a loading network has really helped. Artificial cable networks can be designed and inserted in 1/4 mile increments. We have quite a few users using Asterisk directly connected to station lines or incoming selectors with effective loop lengths of a few feet have used artificial cable networks of 1 to 3 miles with great results. Needless to say, this requires an understanding of basic telephony, some electrical engineering and use of stable capacitors and inductors which may be beyond the ability of many. Country and existing loop length, type of cable loading, if any, are all variables to take into consideration, as well as the specific card involved. the condition certainly can be improved, even with the old X100 card! John Novack -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording calls again
On Wed, Jul 29, 2009 at 5:25 PM, Mr. Rodriguezclubtorr...@hotmail.com wrote: They excuse my question again. I explain to them. I have an equipment 3com v3000 with 20 extensions, my question is if I can use asterisk to record the calls, is necessary an additional servant? at the moment only I have the telephones connected to this equipment. Where encounter information of how doing it? Yes, with asterisk it is possible to record calls. I suggest reading http://www.voip-info.org/wiki/view/MixMonitor I do not understand the server you have, so I do not know whether it will be adequate for recording 20 extensions. The answer is probably yes. Many people on this list record a LOT of calls. One suggestion is that if you have to record many simultaneous recording is to use a ramdisk and periodically synchronize the recordings to disk. This reduces simultaneous threads that have to write to a physical disk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP vs Analog lines
In the last couple of years I can only think of two sites where we've installed E1 connections, compared to many tens of sites where we've gone with VoIP only. You can mitigate against QoS issues by simply installing two DSL connections (one for internet, one for voice). With a decent load balancing router (pfSense works well) this also gives your users failover - if one DSL fails, both internet and voice continue to function. Sure, call quality might suffer and internet speed may drop off a little, but both will continue to work. In fact, one of our sites (a large international company just outside Coventry) has two E1s and a number of IP connections (both fibre and ADSL), and the IP links have been noticeably more reliable than the BT-provided E1s. I don't know what the story is with reliable IP links in other countries, but I agree with Gordon's comments in relation to the UK: Here in the UK we have a generally excellent wholesale (ADSL) broadband system, let-down at times by the back-end ISPs who buy into it, so picking a good back-end ISP is worth it. There's a significant advantage to be gained by peering directly with your suppliers. For example, we peer with our ADSL wholesalers, as well as with our upstream providers (both for incoming and outbound calls). This prevents most calls from going over an IP link where we (by which I mean us and our suppliers, collectively) do not have end-to-end control. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC VPM ?
Noah Miller wrote: Next question: does anybody know how to handle extremely long tail echo that a VPM module cannot? How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can handle 128ms echo tails, which is pretty darn long. It's rare to see an echo tail longer than that except on very high latency connections, or when the echo is actually acoustically generated by the far end and not by network effects. I haven't done any real measurement on it, but I believe it's actually longer than 128ms. As I go higher and higher with echocancel values, the echo does get better, but is never totally eliminated. At echocancel=1024, there is still rather pronounced echo on calls in the local exchanges. The calls are also more or less half-duplex at that point because the vpm is filtering out so much of the signal as echo. Tails are *never* long. Nobody really builds a 128ms canceller. They cancel a few specific sections of delay within a 128ms interval. Typically four 8ms sections - one for each end of the local analogue link, and one for each end of the far analogue link. Things should not get progressively better by gradually lengthening the echo canceller. The echo should basically just disappear when the length is sufficient, and the algorithm can place and tune an 8ms canceller at the right delay. I may just tell the client to look at a partial PRI. All this echo chasing is getting costly for them. If the echo varies with which number you call, its predominantly from the far end. In those cases PRI won't help. You still need to cancel that far end echo. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk Inter digit delay
On Tue, Jul 28, 2009 at 1:01 AM, hadi motamedimotamed...@gmail.com wrote: Dear All Can you please let us know how we can modify our Asterisk inter digit delay ? Actually , our subs dials his intended numbers with some delay in between entering the digits sequentially . It seems that our Asterisk pbx will wait for about 2 seconds and if no extra digits are to be entered then he will decide on routing the dialed number or play the related anouncement . For our current application , it seems that this amount of delay is a little bit small and so please let us know how we can incraese this amount of delay to say 4 seconds . Delay between what? Is this a phone being dialed with a dial tone? Is this an IVR prompt timeout? If you do a Read(), do a longer timeout. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Pavilion at AstriCon: Your project wanted!
John Todd wrote: What your project should have: - No significant corporate sponsorship JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ Isn't that requirement a little hypocritical since Asterisk is heavily corporate sponsored? Just asking, Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not getting inbound CallerID name on Asterisk
On Sun, Jul 26, 2009 at 1:19 PM, Chris Douglaschris.doug...@pioneerballoon.com wrote: We have an inbound PRI connected to our Cisco 3825 router which is then passing the calls to Asterisk as SIP calls. We're getting the CallerID number but not the CallerID name. We are seeing the name in the RPID field with a SIP trace on the Asterisk box but don't understand why it's not registering as the CallerID name. What do you get when you enable debugging on the asterisk cli? core set verbose 3 make a call. Do you see caller ID going through? What does your dialplan look like? You can use NoOp() calls to pop out values including a caller ID if it exists. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for wisdom - One Asterisk system - Multi-incoming trunks
I'm pretty new to this whole Asterisk system VoIP thing, but being a programmer by trade the complexity didn't scare me off (at least not yet)... I have setup an Asterisk system for my home home office. My wife I run two separate businesses from home, and we have a general family home phone line as well. The cost of all these lines with analog carriers was getting ridiculous, so I'm moving over to a SIP carrier. I created one account for a single phone number with a SIP carrier (BroadVoice) and have it working well with my Asterisk system and one SIP phone here as a test. I have IPCop as my Firewall/IDS system and all the SIP/NAT routing stuff is working fine (now). I started the process today to get our other phone numbers moved over to BroadVoice. I checked with them regarding how this is setup and they said that what I was doing was ok, but I thought getting some 'peer review' on this wouldn't be such a bad idea so I welcome any comments, etc. on this. My approach is to have one trunk provided by the SIP provider. All numbers are allocated to that trunk (BroadVoice let me do that when I setup the number transfer). Asterisk receives an incoming call on that trunk and determines the calling number that it was requesting (not sure how to get this, but Broadvoice assured me I could). Anyway after determining what the call was destined for, I then route the call to the appropriate context in the extensions to handle it. I'm fine with setting up all the logic, flow, etc. for the calls. But here's where I'm not sure what to do. I'm getting 4 line Grandstream phones for my office and my wife's office. And an ATA adapter for the general home line. The home line will always call out using the home phone number. The office numbers, however, should change their caller ID and caller name based on which extension is pressed on the phone for the outgoing call. I can see how to do this with the Grandstream SIP phones, and have this working ok for my test phone line. Broadvoice, however, won't let me change the outgoing caller ID. Apparently they have to do this on a trunk by trunk basis. So if I want to have an outgoing call go through line 1 (let's say its ACME Inc), I want it to show 'XXX-XXX- Acme Inc' for the Caller ID. But if the call is being sent through line 2 (let's say its SMITH PROPERTY) I want it to show 'YYY-YYY- Smith Property' for the Caller ID. It looks like in order to do that, I need to purchase separate trunks for each of the outgoing lines. Does this sound right? Should I have purchased all separate trunks up front and then have the phone number transfer associated with the trunk for it? Or is this only something that will affect outgoing calls, so its not a big deal? And what about when the line is busy? How is that handled? I was on the phone yesterday when another call came in, and it came in, jumped to a different extension and then eventually went to voice mail as I didn't answer it. Will my plan to use one trunk for all incoming lines make sense here, or am I likely to get all of this mixed up with calls coming in for one business and being routed to the wrong place? Any suggestions, thoughts or critique would be greatly appreciated. Thank you wise Asterisk gurus! Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for wisdom - One Asterisk system - Multi-incoming trunks
On Wed, 29 Jul 2009, Myles Wakeham wrote: I'm pretty new to this whole Asterisk system VoIP thing, but being a programmer by trade the complexity didn't scare me off (at least not yet)... I have setup an Asterisk system for my home home office. My wife I run two separate businesses from home, and we have a general family home phone line as well. The cost of all these lines with analog carriers was getting ridiculous, so I'm moving over to a SIP carrier. I created one account for a single phone number with a SIP carrier (BroadVoice) and have it working well with my Asterisk system and one SIP phone here as a test. I have IPCop as my Firewall/IDS system and all the SIP/NAT routing stuff is working fine (now). I started the process today to get our other phone numbers moved over to BroadVoice. I checked with them regarding how this is setup and they said that what I was doing was ok, but I thought getting some 'peer review' on this wouldn't be such a bad idea so I welcome any comments, etc. on this. My approach is to have one trunk provided by the SIP provider. All numbers are allocated to that trunk (BroadVoice let me do that when I setup the number transfer). Asterisk receives an incoming call on that trunk and determines the calling number that it was requesting (not sure how to get this, but Broadvoice assured me I could). Anyway after determining what the call was destined for, I then route the call to the appropriate context in the extensions to handle it. I'm fine with setting up all the logic, flow, etc. for the calls. But here's where I'm not sure what to do. I'm getting 4 line Grandstream phones for my office and my wife's office. And an ATA adapter for the general home line. The home line will always call out using the home phone number. The office numbers, however, should change their caller ID and caller name based on which extension is pressed on the phone for the outgoing call. I can see how to do this with the Grandstream SIP phones, and have this working ok for my test phone line. Broadvoice, however, won't let me change the outgoing caller ID. Apparently they have to do this on a trunk by trunk basis. So if I want to have an outgoing call go through line 1 (let's say its ACME Inc), I want it to show 'XXX-XXX- Acme Inc' for the Caller ID. But if the call is being sent through line 2 (let's say its SMITH PROPERTY) I want it to show 'YYY-YYY- Smith Property' for the Caller ID. It looks like in order to do that, I need to purchase separate trunks for each of the outgoing lines. Does this sound right? Should I have purchased all separate trunks up front and then have the phone number transfer associated with the trunk for it? Or is this only something that will affect outgoing calls, so its not a big deal? And what about when the line is busy? How is that handled? I was on the phone yesterday when another call came in, and it came in, jumped to a different extension and then eventually went to voice mail as I didn't answer it. Will my plan to use one trunk for all incoming lines make sense here, or am I likely to get all of this mixed up with calls coming in for one business and being routed to the wrong place? Any suggestions, thoughts or critique would be greatly appreciated. Thank you wise Asterisk gurus! Myles Hi Myles, You don't have to send the traffic back to broadvoice for outbound if you don't want or need to. Perhaps you can send the home traffic to Broadvoice and pick another carrier to send your other outbound traffic to, perhaps one that won't be so picky about your outbound CID. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Pavilion at AstriCon: Your project wanted!
Not when you consider that there are plenty of spaces for corporate projects as well. On Wed, Jul 29, 2009 at 9:21 PM, Anthony antho...@rockynet.com wrote: John Todd wrote: What your project should have: - No significant corporate sponsorship JT --- John Todd email:jt...@digium.comemail%3ajt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ Isn't that requirement a little hypocritical since Asterisk is heavily corporate sponsored? Just asking, Anthony ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for wisdom - One Asterisk system - Multi-incoming trunks
On Wed, 29 Jul 2009, Myles Wakeham wrote:
I have setup an Asterisk system for my home home office.
[snip]
The cost of all these lines with analog carriers was getting ridiculous,
so I'm moving over to a SIP carrier. I created one account for a single
phone number with a SIP carrier (BroadVoice)
[snip]
I've never used BroadVoice, so I have nothing good or bad to say about
them. I've used Vitelity.net for several years and am pleased with them.
I have a nominal monthly fee, pay per minute account. They get $1.49 a
month for a DID and $0.0144 per minute. You'd have to use about 2,600
minutes (about 44 hours) before it would cost as much as a $40 per month
analog. They have an unlimited inbound for $7.95 a month.
I started the process today to get our other phone numbers moved over to
BroadVoice.
[snip]
Vitelity.net charges $18 per number ported. I've never done this.
My approach is to have one trunk provided by the SIP provider. All
numbers are allocated to that trunk (BroadVoice let me do that when I
setup the number transfer). Asterisk receives an incoming call on that
trunk and determines the calling number that it was requesting (not sure
how to get this, but Broadvoice assured me I could). Anyway after
determining what the call was destined for, I then route the call to the
appropriate context in the extensions to handle it.
The calls should be delivered with the DID (aka DNIS, DDI, etc). Usually
you pick this up as the ${EXTEN} in your dialplan and go from there.
[snip]
Broadvoice, however, won't let me change the outgoing caller ID.
Apparently they have to do this on a trunk by trunk basis. So if I want
to have an outgoing call go through line 1 (let's say its ACME Inc), I
want it to show 'XXX-XXX- Acme Inc' for the Caller ID.
[snip]
Being able to specify the caller ID number depends on the carrier.
Vitelity.net does. Specifying the caller ID name is not going to work. The
way it works (from 40,000 feet) is that the name is not passed onto the
real telephone system. The carrier for the dialed number looks up the
number in a database and presents that to the dialed number. If you dial
another VOIP account (sip:john-sm...@example.com) your caller ID name
should be passed.
Does this sound right? Should I have purchased all separate trunks up
front and then have the phone number transfer associated with the trunk
for it? Or is this only something that will affect outgoing calls, so
its not a big deal? And what about when the line is busy? How is that
handled? I was on the phone yesterday when another call came in, and it
came in, jumped to a different extension and then eventually went to
voice mail as I didn't answer it. Will my plan to use one trunk for all
incoming lines make sense here, or am I likely to get all of this mixed
up with calls coming in for one business and being routed to the wrong
place?
I'm more comfortable with the word account than trunk. You can have
multiple DIDs numbers associated with the same account. Some providers
make you specify (via their web site) where you want the calls to go. Some
make you configure your Asterisk server so it registers with their
server. I prefer registration because it let's me change things around
easier.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] CIsco 7960 + asterisk: hepl needed
Are there any other phones registered, or is it just this phone that is having issues? The first thing that I see is the qualify=200 line, and I have not had good experience with Cisco devices and any qualify setting. I would try leaving that out. I also have double quotes around the line1_* parameters. See my comments inline. On Tue, Jul 28, 2009 at 2:14 PM, pepesz76 pepes...@o2.pl wrote: Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing 55 phone icon with x so it looks like the phone is not registered. The phone and the asterisk are in the same local network. On asterisk side: Cawdor*CLI sip show peers ... 55/55 (Unspecified)D N 5060 UNKNOWN ... sip.conf: [55] type=friend defaultuser=55 secret=12345655 context=home_castle callerid=Lukasz Cisco 7960 55 canreinvite=no host=dynamic dtmfmode=rfc2833 Remove: qualify=200 Add: disallow=all allow=ulaw (Or whatever codecs you are using) buggymwi=yes SIPDefault.cnf: image_version: P0S3-8-12-00 proxy1_address: 192.168.1.109 ; Can be dotted IP or FQDN proxy_register: 1 messages_uri: 80 phone_password: cisco ; Limited to 31 characters (Default - cisco) sntp_mode: directedbroadcast ;unicast sntp_server: 192.168.1.77 time_zone: GMT+01/00 ; assuming you're in GMT time_format_24hr: 1 ; to show the time in 24hour format date_format: D/M/Y ; format you would like the date in dial_template: dialplan SIPMAC.cnf: image_version: P0S3-8-12-00 line1_name: 55 line1_name: 55 line1_authname: 55 line1_password: 12345655 line1_shortname: 55 line1_displayname: Lukasz Cisco7960 line1_authname: 55 line1_shortname: 55 ; displayed on the phones softkey line1_password: 12345655 line1_displayname: Lukasz Cisco7960; the caller id proxy1_port: 5060 proxy1_address: 192.168.1.109 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Castle ; add a space at the end, looks neater phone_password: cisco ; Limited to 31 characters (Default - cisco) user_info: none telnet_level: 2 If that still doesn't work, then telnet into the phone and see what is going on. Commands like show config show register etch are very useful for this kind of troubleshooting. If the phone was attached to a CallManager using SIP before, then there could be some bad configuration still stuck in the phone. If you don't specify a new value, these phones cache the old config. Try factory defaulting the phone if all else fails. I have quite a few of these phones working without issue. Good luck! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk Inter digit delay
Dear David I mean when the subs dials the digits with some delay between entering the digits sequentially . At our current case , the Asterisk will wait about 2 seconds to see if another digit will be dialed or not and then he will route the dialed digits according to the pre-defined routing table or he will play the appropriate announcements . We are expected to increase this inter digit delay to say 4 seconds . Please let us know how we can increase this parameter in our Asterisk configuration files . Regards H.Motamedi On Thu, Jul 30, 2009 at 3:19 AM, David Backeberg dbackeb...@gmail.comwrote: On Tue, Jul 28, 2009 at 1:01 AM, hadi motamedimotamed...@gmail.com wrote: Dear All Can you please let us know how we can modify our Asterisk inter digit delay ? Actually , our subs dials his intended numbers with some delay in between entering the digits sequentially . It seems that our Asterisk pbx will wait for about 2 seconds and if no extra digits are to be entered then he will decide on routing the dialed number or play the related anouncement . For our current application , it seems that this amount of delay is a little bit small and so please let us know how we can incraese this amount of delay to say 4 seconds . Delay between what? Is this a phone being dialed with a dial tone? Is this an IVR prompt timeout? If you do a Read(), do a longer timeout. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Out of office
Thank-you for your email. I will be out of the office from Thursday, July 30 until Monday, August 3. I will have limited access to email during this time and will respond as soon as possible. If you need immediate assistance, please call our support line at 770-674-3900 x 1. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk Inter digit delay
I mean when the subs dials the digits with some delay between entering the digits sequentially . At our current case , the Asterisk will wait about 2 seconds to see if another digit will be dialed or not and then he will route the dialed digits according to the pre-defined routing table or he will play the appropriate announcements . We are expected to increase this inter digit delay to say 4 seconds . Please let us know how we can increase this parameter in our Asterisk configuration files . see http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout also, I suggest you read this book : http://astbook.asteriskdocs.org/ hth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk Inter digit delay
Thank you very much for your reply . But please be informed that our current
line-outgoing route is being configured as the followings (in
extensions.conf):
[line-outgoing]
exten = _X.,1,macro(dialuser,Zap/g1/${EXTEN},${EXTEN})
As you see , it is trying to consider the dialed number as an whole packet
(but not based on one-by-one digit basis) . Can you please do us favor and
let us know how we can get benefit of your proposed DigitTimeout command to
fulfill the job ?
Regards
H.Motamedi
On Thu, Jul 30, 2009 at 5:28 AM, Marc Charbonneau
timebandit...@gmail.comwrote:
I mean when the subs dials the digits with some delay between entering
the
digits sequentially . At our current case , the Asterisk will wait about
2
seconds to see if another digit will be dialed or not and then he will
route
the dialed digits according to the pre-defined routing table or he will
play
the appropriate announcements . We are expected to increase this inter
digit
delay to say 4 seconds . Please let us know how we can increase this
parameter in our Asterisk configuration files .
see http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout
also, I suggest you read this book : http://astbook.asteriskdocs.org/
hth
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