[asterisk-users] Call Forwarding / Follow-Me on PRI
Friends, Curious if others have run into this scenario, and can shed further light on it. I am working with an installed base of systems using PRI circuits from several carriers, and the symptoms I relate occur across the board. Most carriers are restricting CALLING Number ID to be one of the numbers allocated to the associated circuit. This makes sense from a perspective of call-fraud prevention. We have clients that used call forwarding or follow-me extensively, and configured to send the ORIGINAL callerID as the Calling ID, so when the call shows up on their cell phone, it appears to be coming from the originator. This capability seems to be going away. Some legacy PRI carriers were not (and perhaps continue to be) so strict about it, and a recent client who was used to this feature, is now unhappy that their new PRI carrier does not allow any callingID other than one associated with the PRI. A workaround or RNIE (Redirecting Number Information Element) has been recommended as an alternative, but that does not appear to be standardized, nor implemented in asterisk. The only PBX vendors that appear to support this in even a limited sense are Cisco and Shoretel. I'm curious if others have encountered this same situation (I'm sure you have), or any other pertinent thoughts. Thanks in advance! -- Barry D. Hassler President, HCST http://www.hcst.com/ 937-427-9000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID (name) - where does it come from?
Well, Teliax says they "have no access to the PSTN's database", but I'm suggesting they check out TargusInfo as mentioned above. One of their suggestions, is to contact the local ILEC to get the number published in their white pages. Will that accomplish the same thing (I doubt it). On Wed, Jul 8, 2009 at 8:51 AM, Danny Nicholas wrote: > CALLERID(name) is a TELCO specific field. In the long run, you will be > best served using your own lookup of a database using CALLERID(num), since > CID(name) is unreliable and in some cases costly. IMO, you would be well > served with an app (AGI?) that recorded valid names into the database and > let you insert the names where they aren’t. > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Barry D. Hassler > *Sent:* Tuesday, July 07, 2009 12:41 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] Caller ID (name) - where does it come from? > > > > Hi Folks, having an issue with outbound calls through a VOIP provider. > Calls get sent out with the CallerID(number), but where does callerID(name) > come from? Apparently not from provider, as we are seeing different > (sometime missing) names on inbound calls, different than what we have > configured. Apparently this comes from some telco database somewhere? > Numbers were ported from a wired-telco. > > > > -- > Barry D. Hassler > President, HCST > > http://www.hcst.net/ > 937-427-9000 > > ___ > -- 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 > -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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] Caller ID (name) - where does it come from?
So how does Teliax (for instance) go about getting their client's information into these directories? Do they establish a relationship with someone like TargusInfo (described above)? How do other ITSP's provide this service, or do they ignore it as well? On Tue, Jul 7, 2009 at 9:49 PM, Frank Bulk wrote: > How does that work? > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro > Sent: Tuesday, July 07, 2009 8:18 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Caller ID (name) - where does it come from? > > > > I get paid every time I call someone that subscribes to caller ID. > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > ___ > -- 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 > > > ___ > -- 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 > -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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] Caller ID (name) - where does it come from?
This is all excellent information. My primary issue is for calls that are
placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of
those calls are the ones that are not getting the proper CNAM information as
the call comes in.
We just recently ported the client's POTS lines to VOIP, and with the
exception of this issue, all is working well. But, my client is really
unhappy that their callerID NAME isn't showing up.
On Tue, Jul 7, 2009 at 3:42 PM, Frank Bulk wrote:
> There's a bit of oversimplification going on here -- it's not "a ...
> database". Different CNAM providers have different databases which are
> populated from many sources. Most of the data probably matches, but not
> all
> of it.
>
> If the Calling Name is incorrect, the person who received the call will
> have
> to check with their telephony provider (or, if they do their own CNAM
> lookups, with their CNAM provider) to get the name for the calling party
> fixed up (this presumes that the calling party has already verified with
> their own telephony provider that their name is correctly listed). But
> that's not all of it, either, because the next time the CNAM provider
> refreshes their records, the local fix could be overridden (I'm not sure if
> any CNAM providers have the capability to ignore old/bad data for a record,
> but perhaps so). Ideally the CNAM provider shares with the calling party
> which database the CNAM provider is using for the calling party, so that
> the
> calling party can try to get it fixed directly with the database provider
> (if that's even possible).
>
> In short, it's a mess.
>
> But because accuracy rates are one of the elements that CNAM providers
> compete on, these usually do get cleaned up.
>
> Frank
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
> Sent: Tuesday, July 07, 2009 1:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?
>
> The Caller ID name, "CNAM" is a separate database owned and maintained
> "cooperatively" by the bell operating companies.
>
> Your ITSP is not doing these CNAM lookups for you because they would have
> to
>
> pay the BOC's for the 'dips' into the CNAM database. CNAM is a little cash
> cow that the BOC's are quick to protect. As such CNAM dips may not be
> cached or re-sold as a term service that you must agree to with your CNAM
> provider.
>
> As far as solving your CNAM problem, you would need to either choose an
> ITSP
>
> that will provide you with CNAM data on a per-call basis, OR you need to do
> CNAM dips yourself as I (and many others) do. Beware that some ITSP's
> provide "best-effort" name data culled from various sources. It's not
> always terrible but it's not 'coke' it's more like 'dollar store' cola. :-)
>
> As a call comes in to your dial plan you can populate the CALLERID(name)
> channel variable using the CURL function in your dialplan as so:
> exten =>
> s,n,Set(CALLERID(name)=${CURL(
> http://cnam1.edicentral.net/getcnam?q=C&f=S&dn
> =${CALLERID(num)})})
>
> AND let's not forget the completely separate issue with getting your
> ITSP-provisioned number ENTERED INTO the CNAM database in the first place,
> so people see "Karl Fife" rather than the "city, state" or worse, some
> string of arcane LATA information. There's a solution to this problem too
> but I digress...
>
> I've posted my personal notes below from about 18 months ago when I was
> searchign for CNAM providers:
>
> -Karl
>
> CNAM PROVIDRES:
>
> Metrostat.com
> about 1.5¢ per dip,
> $30 minimum deposit, refundable
> CNAM service not well documented on web site
> A registerd CLEC
>
> Got Name - Out of business?
> 1.5¢ per dip. no minimums, no setup
>
> ClearReach Networks
> .67¢ per dip $200 monthly minimum, resell ok, significant setup fees
>
> 411xml.com
> more expensive than ClearReach.
>
> - Original Message -
> From: Barry D. Hassler
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Sent: Tuesday, July 07, 2009 12:40 PM
> Subject: [asterisk-users] Caller ID (name) - where does it come from?
>
> Hi Folks, having an issue with outbound calls through a VOIP provider.
> Calls
>
> get sent out with the CallerID(number), but where does callerID(name) come
> from? Apparently not from provider, as we are seeing different (sometime
> missing) names on inbound calls, dif
[asterisk-users] Caller ID (name) - where does it come from?
Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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] Polycom Phones start to break up after being up a LONG time
Folks, I haven't paid attention to these responses, sorry! This appears to be an issue primarily on calls with an EXTERNAL leg, I'm fairly certain it's on both inbound and outbound calls. The frequency of the reports from this client are increasing, and although I was intending to do a mass reboot of all the polycoms over the weekend, I have not yet, but will have to do it tonight (2 more reports today). All the external legs are via PRI. The phones are 501 and 601's, running bootrom 3.2.2.0019 and SIP 2.2.0.0047. There are many of these that ARE on POE switched (Cisco), On Fri, Feb 20, 2009 at 12:34 PM, Jeff LaCoursiere wrote: > > You also don't mention if it is internal to internal or if there is an > external leg involved, and if so what type. > > j > > On Fri, 20 Feb 2009, Asterisk Asterisk wrote: > > > That's interesting - I haven't noticed this with any of my installs. What > version of firmware and SIP? > > > > > > > > > > > > From: Barry D. Hassler > > To: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > > Sent: Friday, February 20, 2009 8:41:33 AM > > Subject: [asterisk-users] Polycom Phones start to break up after being up > a LONG time > > > > Has anyone else encountered this? I have a fairly large installation (~50 > phones, almost all Polycom 501's and a handful of 601's. We're running into > a number of phones on which the outbound voice (Polycom phone user doesn't > hear any problems, but the other end does) is breaking up occasionally -- > enough to be noticeable and make you say "what?". In each case, rebooting > the phone has resolved the symptoms, but I'd like to know if there is a > known problem. > > > > most of these phones would be up for several months now (installed this > past summer), and unless there are any power outages, would not be restarted > specifically. > > > > I'm planning on restarting all the phones over the weekend, but as this > is a 24-hour operation, we'd like to avoid interrupting phones at all. > > > > -- > > Barry D. Hassler > > President, HCST > > > > http://www.hcst.net/ > > 937-427-9000 > > > > > > > > > > ___ > -- 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 > -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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
[asterisk-users] Polycom Phones start to break up after being up a LONG time
Has anyone else encountered this? I have a fairly large installation (~50 phones, almost all Polycom 501's and a handful of 601's. We're running into a number of phones on which the outbound voice (Polycom phone user doesn't hear any problems, but the other end does) is breaking up occasionally -- enough to be noticeable and make you say "what?". In each case, rebooting the phone has resolved the symptoms, but I'd like to know if there is a known problem. most of these phones would be up for several months now (installed this past summer), and unless there are any power outages, would not be restarted specifically. I'm planning on restarting all the phones over the weekend, but as this is a 24-hour operation, we'd like to avoid interrupting phones at all. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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
[asterisk-users] Timestamp on voice mail messages is based on wrong timezone
running asterisk 1.4.13... I've noticed with SOME email clients that the timestamp reported from voicemail is 5 hours off (difference of EST vs UTC). That is, a voicemail received at 15:17:59 EST is sent via email with a Date: header of "10:17:59 - 0500". An email sent through normal means (mail client) from the asterisk server reports the correct time, so it's definitely something inside Asterisk that is inserting the wrong Date stamp. Most email clients seem to display messages by the date RECEIVED, as opposed to date SENT, but naturally, I am the silly one who views email by date SENT, so my voicemails get dropped back 5 hours (which in many cases is no longer on the first screen of my email index). Following are some of the applicable email headers showing this situation Notice the time stamps highlighted. X-MimeOLE: Produced By Microsoft Exchange V6.5 Received: from chi.hcst.net ([192.168.254.230]) by hcst.com with Microsoft SMTPSVC(6.0.3790.3959); Tue, 20 Jan 2009 15:14:00 -0500 MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="_=_NextPart_001_01C97B3B.A1867400" Received: from asterisk-bvr.hcst.com (asterisk-bvr.hcst.com [192.52.183.237]) by chi.hcst.net (8.13.6/8.13.6) with ESMTP id n0KKE0kn024700 for ; Tue, 20 Jan 2009 15:14:00 -0500 Received: from asterisk-bvr.hcst.com (localhost [127.0.0.1]) by asterisk-bvr.hcst.com (8.14.0/8.14.0) with ESMTP id n0KKE1jd003654 for ; Tue, 20 Jan 2009 15:14:01 -0500 Received: (from r...@localhost) by asterisk-bvr.hcst.com (8.14.0/8.13.6/Submit) id n0KKE0vX003643; Tue, 20 Jan 2009 15:14:00 -0500 Return-Path: X-OriginalArrivalTime: 20 Jan 2009 20:14:00.0942 (UTC) FILETIME=[A21630E0:01C97B3B] X-Asterisk-CallerID: 2311 X-Asterisk-CallerIDName: Barry D. Hassler Content-class: urn:content-classes:message Subject: [PBX] New message 1 in mailbox 2302 Date: Tue, 20 Jan 2009 10:14:00 -0500 Message-ID: X-MS-Has-Attach: yes X-MS-TNEF-Correlator: Thread-Topic: [PBX] New message 1 in mailbox 2302 thread-index: Acl7O6I+TAESdG+3RlmwhFhRW9rXEg== From: "HCST Asterisk PBX" To: "Barry D. Hassler" X-Evolution-Source: exchange://hass...@zeta.hcst.com/ -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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
[asterisk-users] Digium Asterisk Appliance voicemail & logs
Does anyone know how much space the appliance has for voicemail and/or logs? Doesn't have an embedded disk from what I can see, and only a 1G flash card? -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --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] Asterisk 1.4.13 -- issue with parked calls
We park the calls by transferring to extension 7000, which is our parking extension. We have both Zap and SIP extensions, and I haven't been able to see a pattern if its related to one or the other. The primary person answering the phone is using a SIP phone (Grandstream GXP-2000), we have a small number of analog phones left (2), and other SIP phones (mostly Polycom). The only clue I've seen with the CLI is that I'll generally see a LOT of entries for active channels on extension 7000 if I do a "show channels". I'll try to catch this situation again and grab the output. I only started having this problem when I upgraded to the 1.4 version from 1.2. On 10/31/07, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote: > > Barry D. Hassler wrote: > > I've tried to find other threads with this same topic, but haven't > > found any... Apologies if this already being discussed > > > > Running asterisk 1.4.13 (upgraded from 1.4.9) and zaptel 1.4.4. > > > > Having an issue with (I think) parked calls. We tend to park calls, > > but we're often not able to pick them back up, or the other party says > > they get dropped, etc. There doesn't seem to be a specific pattern > > that I've discovered so far. I had this happen to me personally this > > morning -- receptionist parked a call for me on extension 7001, but > > when I dialed 7001, just got dead air. I could see in asterisk that > > the call was indeed parked though, and after calling the person back, > > he reported he was just hearing the lovely on-hold music. > > > > Is there a known issue (and even better, a fix) for this situation? > > Any other information I can provide I'll do so! > What kind of phones are you using? are they Zap or SIP? > > Can you provide a CLI output with any tips in it? > > > > ___ > --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 > -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --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
[asterisk-users] Asterisk 1.4.13 -- issue with parked calls
I've tried to find other threads with this same topic, but haven't found any... Apologies if this already being discussed Running asterisk 1.4.13 (upgraded from 1.4.9) and zaptel 1.4.4. Having an issue with (I think) parked calls. We tend to park calls, but we're often not able to pick them back up, or the other party says they get dropped, etc. There doesn't seem to be a specific pattern that I've discovered so far. I had this happen to me personally this morning -- receptionist parked a call for me on extension 7001, but when I dialed 7001, just got dead air. I could see in asterisk that the call was indeed parked though, and after calling the person back, he reported he was just hearing the lovely on-hold music. Is there a known issue (and even better, a fix) for this situation? Any other information I can provide I'll do so! -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --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] SIP phone recommendation (used to be: no subject)
I'd go with Polycom all the way. We have a number of different types of phones in use, or that we've worked with, including Grandstream, SIpura and Atacom, and the quality difference with the Polycom phones is astounding. On 10/29/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > My apologies to the list for not having entered a subject line in the > email. > > Thanks > > On Oct 29, 2007, at 1:42 PM, [EMAIL PROTECTED] wrote: > > > Hi all, > > > > We have a client that needs to setup about 80 desk phones (about 50 > > in one location and about another 30 in 5 different locations). Which > > brand/model would you recommend. We were personally thinking in > > recommending either Cisco, Aastra, Polycom, or Snom, for we've heard > > great things about them. However, having no real experience with them > > makes it hard in recommending one to our customer. The only > > experience we've had is a very frustrating one trying to load the IP > > software on a Cisco 7970G and so we assume that if we have to go > > through that for all 80 phones, we'll probably commit suicide :) > > > > Thanks > > > > > > ___ > > --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 > > > ___ > --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 > -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --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] A question about an install i have been asked about...
The point about the "politcal" side of this is extremely valid. But it also becomes a business opportunity to provide the hosted PBX as a service to BOTH companies! Beyond that, some sort of written agreement between the two companies, the one "owning" the box and the other as a client may be worthwhile. On 4/13/07, Gordon Henderson <[EMAIL PROTECTED]> wrote: On Fri, 13 Apr 2007, Gavin Spurgeon wrote: > Hi List... > > Lets say I have been asked to do an Asterisk/Trixbox install for an > environment where 2 companies 'live' in the same... building... > > Both companies have 2 incoming Analogue BT Lines and then have good old > BT Phones plugged into them.. > > I have been asked to setup a VoIP system for one of the companies and > the other company has showed an interest, There is one central Server > Room/Cabinet will all of the shared services (www,file server, etc, > etc...) and I thought of going down the lies of installing one more > Dedicated * Box into this cabinet with would service both companies... > > My 1st questions are as follows > > 1.) can one * box host both separate phone setups and handle them as > separate setups.. (I.e. user in company A can not dial and internal line > to user in company B) > > 2.) can I use a Single 4 FXO port card and reserve 2 ports for one > company and the other 2 ports for the other company? > > This would also need to work for outgoing calls, all calls made by > company 1 MUST go over the 2 lines that belong to company 1 and of > course the same for company 2... > > 3.) How easy is this setup ? or should I just make up 2 different * > Boxes, 1 for each company ? Answers to the above are generally "Yes". People sell hosted VoIP boxes all the time (centrex/virtual PBX), and you can bet it's not one box per client - just like web hosting, that's not really financially viable unless they are paying lots of money! But one issue you might have is the LAN to both companies. A typical * box might only have one Ethernet port - do both companies share the same LAN (I'd generally hope not!) Also the outgoing Internet connections - do they have their own separate ADSL lines, or share one? If they're separate and want to do the remote working thing, then routing the calls to each one is an additional factor. If there's one Internet connection and they have a DMZ (small subnet routed down the ADSL line) and 2 routers, one for each company, then putting the asterisk box in the DMZ is the thing to do, but you then have issues with NAT (probably) and firewalling for the phones on the inside. Not insurmountable, but an additional thing to go wrong. And then there's the "political" side of things - what if one company decided it doesn't like the other company anymore and pulls the plug on the PBX ... Or gets access to the box and listens to all their voicemail? (Although since they already seem to have one common comms cupboard, it would seem they are friendly enough already!) So if you want a GUI to let each company manage their own extensions then you'll have to write your own - and at that point it might be just as easy to install 2 boxes... > I hope someone could shed some light on this for me > and give me some idea's.. Read-up on contexts, and I'd suggest you look into asterisk in the raw rather than rely on a GUI of some sorts... The book helps. Asterisk, The future of Telephony. You can get it in PDF form. Personally I'd install 2 boxes. Gordon ___________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --Bandwidth 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] A question about an install i have been asked about...
The short answer is sure, one * box can handle multiple companies, each with their individual "personalities" This will take some dial-plan coding, which may be difficult in the trixbox environment. There are "GUIs" which proclaim to specifically support this type of environment -- Thirdlane being an instance I've heard of (but have not used). I have installations with this specific environment, but using all custom dialplan coding. On 4/13/07, Gavin Spurgeon <[EMAIL PROTECTED]> wrote: Hi List... Lets say I have been asked to do an Asterisk/Trixbox install for an environment where 2 companies 'live' in the same... building... Both companies have 2 incoming Analogue BT Lines and then have good old BT Phones plugged into them.. I have been asked to setup a VoIP system for one of the companies and the other company has showed an interest, There is one central Server Room/Cabinet will all of the shared services (www,file server, etc, etc...) and I thought of going down the lies of installing one more Dedicated * Box into this cabinet with would service both companies... My 1st questions are as follows 1.) can one * box host both separate phone setups and handle them as separate setups.. (I.e. user in company A can not dial and internal line to user in company B) 2.) can I use a Single 4 FXO port card and reserve 2 ports for one company and the other 2 ports for the other company? This would also need to work for outgoing calls, all calls made by company 1 MUST go over the 2 lines that belong to company 1 and of course the same for company 2... 3.) How easy is this setup ? or should I just make up 2 different * Boxes, 1 for each company ? I hope someone could shed some light on this for me and give me some idea's.. Maybe even point me in the direction of some HOWTOs on making a 'Hosted * Server' type setup. Best Regards Gavin Spurgeon Assistant Systems Administrator Leigh City Technology College [EMAIL PROTECTED] http://www.leighctc.kent.sch.uk Tel: 01322 620501 Fax: 01322 620599 IS HelpDesk : Ext 541 -- This message has been scanned for viruses and dangerous content by the Systems @ the LeighCTC, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call forwarding (from PHONE configuration) with PRI
Hi folks.
My client is wanting to use call forwarding configured on their phones
(Linksys SPA942), with a PRI from their provider. When we configure call
forwarding, we invariably get a "The number you have dialed is not in
service" message from the providers.
Examining the detailed dial plan debugging as well as the PRI debugging,
the number is dialed correctly. The only difference noticed between a
forwarded call versus a "normal" outbound call is that an extended
attribute of "Forwarded Unconditionally".
The (slightly edited -- I've edited the phone numbers to protect the
innocent) trace is below. Is this an asterisk issue, or an ILEC issue? I
have a ticket open with the ILEC as well, but they tend to be less than
helpful (their response has been to turn on their call forwarding option).
Any thoughts appreciated!
[ 02 01 01 1e ]
-- Got SIP response 302 "Moved Temporarily" back from 192.168.1.55
mail*CLI>
-- Now forwarding Zap/1-1 to
'Local/[EMAIL PROTECTED]' (thanks to SIP/2105-08217980)
-- Executing Macro("Local/[EMAIL PROTECTED],2",
"to-pstn-standard|1937000") in new stack
-- Executing NoOp("Local/[EMAIL PROTECTED],2",
""CALLERID(num)=" 937001") in new stack
-- Executing NoOp("Local/[EMAIL PROTECTED],2",
""MACRO_EXTEN=" 1937000") in new stack
-- Executing GotoIf("Local/[EMAIL PROTECTED],2",
"0?:7") in new stack
-- Goto (macro-to-pstn-standard,s,7)
-- Executing NoOp("Local/[EMAIL PROTECTED],2", ""Call
Forwarded"") in new stack
-- Executing Set("Local/[EMAIL PROTECTED],2",
"CALLERID(name)="CLIENT NAME"") in new stack
-- Executing Set("Local/[EMAIL PROTECTED],2",
"CALLERID(num)=513001") in new stack
-- Executing Dial("Local/[EMAIL PROTECTED],2",
"Zap/g1/1937000||r") in new stack
-- Requested transfer capability: 0x00 - SPEECH
mail*CLI>
[04 03 80 90 a2]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps,
circuit-mode (16)
Ext: 1 User information layer 1: u-Law
(34)
[18 03 a9 83 82]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive
Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel
Type: 3
Ext: 1 Channel: 2 ]
[1e 02 80 83]
Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0)
0: 0 Location: User (0)
Ext: 1 Progress Description: Calling
equipment is non-ISDN. (3) ]
[28 19 b1 22 46 57 44 20 46 52 4f 4d 20 42 4c 55 45 20 4c 4f 4f 50 20
4c 4c 43 22]
Display (len=25) Charset: 31 [ "FWD FROM x" ]
[6c 0c 49 80 35 31 33 32 30 34 32 31 30 30]
Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI:
Private Numbering Plan (9)
Presentation: Presentation permitted, user
number not screened (0) '513001' ]
[70 0c c9 31 39 33 37 34 32 37 39 30 30 30]
Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI:
Private Numbering Plan (9) '1937000' ]
[74 07 49 01 8f 32 31 30 35]
Redirecting Number (len= 9) [ Ext: 0 TON: Subscriber Number (4) NPI:
Private Numbering Plan (9)
Ext: 0 Presentation: Presentation
permitted, user number passed network screening (1)
Ext: 1 Reason: Forwarded unconditionally
(15) '2105' ]
-- Called g1/1937000
-- Local/[EMAIL PROTECTED],1 is ringing
mail*CLI>
--
Barry D. Hassler
President, HCST
http://www.hcst.net/
937-427-9000
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Re: [asterisk-users] Re-parking (or transfer) a parked call
Thanks Marc, hadn't seen that one. I'm currently at 1.2.14, looks like this is in .15. Thank you very much! On 3/12/07, Marc Archer <[EMAIL PROTECTED]> wrote: Barry, Have a look at http://bugs.digium.com/view.php?id=8804 I am assuming that you are trying to transfer using the # key (or whatever is specified in features.conf) to re-park or transfer the parked call. I think this has been fixed in the latest versions of Asterisk 1.2 Marc. *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Barry D. Hassler *Sent:* Monday, 12 March 2007 3:13 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Re-parking (or transfer) a parked call How do you transfer or re-park a call that's been picked up from a parking lot? I don't see any options for specifying the transfer options on the parked call, so that you could transfer or repark it. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re-parking (or transfer) a parked call
How do you transfer or re-park a call that's been picked up from a parking lot? I don't see any options for specifying the transfer options on the parked call, so that you could transfer or repark it. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ --Bandwidth 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] Caller ID not getting to analog extensions
There are 3 or 4 analog phones connected on the FXS port. Only 2 of them have callerID. On the CNAME as opposed to CNID, have NO idea! The callerID worked fine on these phones until I cut them over to the asterisk server this weekend. On 2/26/07, Jay R. Ashworth <[EMAIL PROTECTED]> wrote: On Mon, Feb 26, 2007 at 04:31:07PM -0500, Barry D. Hassler wrote: >Recent installation with a simple TDM11B (one FXO, one FXS) that >I've set up (at home). I am receiving callerID fine from the telco, >as it shows up in my call detail records, AND on 2 SIP phones. >However, I'm not reliably receiving it (that is, very seldom does >it come through) on the analog phones. Any ideas on where to check >configurations, etc? I haven't encountered this issue before (my >other installations are always much larger than this one for home). Two hipshots: How *many* analog phones on your one FXS? And is it possible that the system is sending CNAME, not just CNID, and the phones don't do names, and are confused? Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth & AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Barry D. Hassler ___ --Bandwidth 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] Caller ID not getting to analog extensions
Hi Folks, Recent installation with a simple TDM11B (one FXO, one FXS) that I've set up (at home). I am receiving callerID fine from the telco, as it shows up in my call detail records, AND on 2 SIP phones. However, I'm not reliably receiving it (that is, very seldom does it come through) on the analog phones. Any ideas on where to check configurations, etc? I haven't encountered this issue before (my other installations are always much larger than this one for home). -- Barry D. Hassler ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Rhino cards lock up system -- anyone else ever seen this?
Turns out this appears to be related to hald -- the hardware abstraction layer daemon running on Centos. I had the almost identical situation occur with a completely separate system which I loaded Trixbox up on, with a single Digium TDM400P card in it. Struggled for several hours over the weekend trying to figure it out. I ended up shutting off all the services I didn't specifically need, and turned them back on one at a time (turned out, hald was the first I tried -- I was most suspicious with it). As soon as I started it up, it locked the system up. Turned all the other services back on, leaving hald off, and the system is running fine. Did the same with the original problem system, and now have no problems with the Rhino cards either! On 1/23/07, Barry D. Hassler <[EMAIL PROTECTED]> wrote: Hi Folks, Struggling with a new * installation with 2 Rhino R2T1 cards. For some reason, the system is locking up tight when you run ztcfg to configure the card(s). Configuration is asterisk 1.2.14, zaptel 1.2.12, and rhino's 1.05rxt1 drivers. The cards seem to load fine with a "modprobe rxt1", but once you run "ztcfg -vvv", the system will lock up within a few seconds, no errors reported in logs or console. I'm stumped, Rhino is stumped, and I haven't seen any other threads of this nature. -- Barry D. Hassler -- Barry D. Hassler ___ --Bandwidth 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] Rhino cards lock up system -- anyone else ever seen this?
Hi Folks, Struggling with a new * installation with 2 Rhino R2T1 cards. For some reason, the system is locking up tight when you run ztcfg to configure the card(s). Configuration is asterisk 1.2.14, zaptel 1.2.12, and rhino's 1.05rxt1 drivers. The cards seem to load fine with a "modprobe rxt1", but once you run "ztcfg -vvv", the system will lock up within a few seconds, no errors reported in logs or console. I'm stumped, Rhino is stumped, and I haven't seen any other threads of this nature. -- Barry D. Hassler ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to play pre-recorded file in meetme conference
GOT IT !!! Here's what I was trying, using the admin interface... Action: Originate Channel: Local/8020 Application: Playback Data: hcst/intro I have 8020 set up as a meetme conference (and had 2 other legs connected), and hcst/intro is a prerecorded file. All I was missing was the context on the Channel: Channel: Local/[EMAIL PROTECTED] It works! On Thu, 2006-10-12 at 19:18 +0100, Tim Panton wrote: On 10 Oct 2006, at 22:33, Barry D. Hassler wrote: > I was playing around with that idea myself, but I can't find a way > to place the call which will actually play the recording. What I'd > like to accomplish is that somewhere in a conference call, I'd like > to be able to say "let me play this recording for you...", and > somehow initiate it. There is no "Channel" to dial Ah, you can use the Local channel for this sort of thing. Tim Panton www.mexuar.com Barry D. Hassler President HCST 2332 Grange Hall Road Beavercreek, Ohio 45431-2345 http://www.hcst.net/ [EMAIL PROTECTED] +1 937-427-9000 +1 937-427-8706 FAX FWD: 3934279000 (655480) HCST*Net Support Issues: please email [EMAIL PROTECTED] Billing Issues: Please email [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to play pre-recorded file in meetme conference
I was playing around with that idea myself, but I can't find a way to place the call which will actually play the recording. What I'd like to accomplish is that somewhere in a conference call, I'd like to be able to say "let me play this recording for you...", and somehow initiate it. There is no "Channel" to dial Maybe it's not possible, but sure am trying it! On Tue, 2006-10-10 at 07:09 -0400, Brian Rogan wrote: I don't know if there is a better way to do this with meetme itself, but you could use the manager interface (or even the file method described in http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out). You can pass a Data argument with the filename, to an extension that simply plays a file into the conference. You may also be able to do something with the 'b' argument to MeetMe. --Brian On Mon, Oct 09, 2006 at 04:42:02PM -0400, Barry D. Hassler wrote: > Hey folks, Is it possible to play a pre-recorded file in a meetme > conference? That is, I'd like to get everyone into a conference, then > somehow play a previously recorded file (in this case, a podcast). This > isn't for individuals to call into to listen to the cast, but for it to > be played simultaneously for all in the conference. > > This would be handy for me! > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to play pre-recorded file in meetme conference
Hey folks, Is it possible to play a pre-recorded file in a meetme conference? That is, I'd like to get everyone into a conference, then somehow play a previously recorded file (in this case, a podcast). This isn't for individuals to call into to listen to the cast, but for it to be played simultaneously for all in the conference. This would be handy for me! ___ --Bandwidth 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] Spurious hangups on zaptel interface
Title: RE: [asterisk-users] Spurious hangups on zaptel interface Commenting out the busydetect=yes seems to have resolved this annoying issue! Thanks Eric! On Thu, 2006-09-28 at 02:26 -0400, Barry D. Hassler wrote: I did have busydetect=yes in my config, but not the callprogress.I've commented out busydetect, and we'll try some of these same calls to see what happens. -Original Message- From: Eric "ManxPower" Wieling [mailto:[EMAIL PROTECTED]] Sent: Wed 9/27/2006 6:51 PM To: Barry D. Hassler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spurious hangups on zaptel interface Barry D. Hassler wrote: > We seem to be getting unexpected hangups on our * system, very > consistent when calling particular numbers that we can associate with a > clients phone system. These hangups generally occur when our call is > transferred within their system (to voicemail usually). > > I'm suspecting their may be some sort of "flash" (for lack of a better > term) on the called side, but I can't verify this. > > the situation does appear to be consistent and reproducible, but only > with specific phone systems that our calls go through. > > Has anyone else experienced this, or have any potential resolutions? > I've researched this quite a bit, but not turning up anything > particularly relevant. > > I am using asterisk 1.2.9.1 Remove busydetect=yes and callprogress=yes from your /etc/asterisk/zapata.conf Barry D. Hassler President HCST 2332 Grange Hall Road Beavercreek, Ohio 45431-2345 http://www.hcst.net/ [EMAIL PROTECTED] +1 937-427-9000 +1 937-427-8706 FAX FWD: 3934279000 (655480) HCST*Net Support Issues: please email [EMAIL PROTECTED] Billing Issues: Please email [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Spurious hangups on zaptel interface
Title: RE: [asterisk-users] Spurious hangups on zaptel interface I did have busydetect=yes in my config, but not the callprogress.I've commented out busydetect, and we'll try some of these same calls to see what happens. -Original Message- From: Eric "ManxPower" Wieling [mailto:[EMAIL PROTECTED]] Sent: Wed 9/27/2006 6:51 PM To: Barry D. Hassler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spurious hangups on zaptel interface Barry D. Hassler wrote: > We seem to be getting unexpected hangups on our * system, very > consistent when calling particular numbers that we can associate with a > clients phone system. These hangups generally occur when our call is > transferred within their system (to voicemail usually). > > I'm suspecting their may be some sort of "flash" (for lack of a better > term) on the called side, but I can't verify this. > > the situation does appear to be consistent and reproducible, but only > with specific phone systems that our calls go through. > > Has anyone else experienced this, or have any potential resolutions? > I've researched this quite a bit, but not turning up anything > particularly relevant. > > I am using asterisk 1.2.9.1 Remove busydetect=yes and callprogress=yes from your /etc/asterisk/zapata.conf ___ --Bandwidth 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] Spurious hangups on zaptel interface
We seem to be getting unexpected hangups on our * system, very consistent when calling particular numbers that we can associate with a clients phone system. These hangups generally occur when our call is transferred within their system (to voicemail usually). I'm suspecting their may be some sort of "flash" (for lack of a better term) on the called side, but I can't verify this. the situation does appear to be consistent and reproducible, but only with specific phone systems that our calls go through. Has anyone else experienced this, or have any potential resolutions? I've researched this quite a bit, but not turning up anything particularly relevant. I am using asterisk 1.2.9.1 Barry D. Hassler President HCST 2332 Grange Hall Road Beavercreek, Ohio 45431-2345 http://www.hcst.net/ [EMAIL PROTECTED] +1 937-427-9000 +1 937-427-8706 FAX FWD: 3934279000 (655480) HCST*Net Support Issues: please email [EMAIL PROTECTED] Billing Issues: Please email [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ZAP: psuedo camped on channel 1?
Folks, this issue has been plaguing me for about 2 months now, ever since I did an upgrade of my asterisk system (new hardware and software -- updated OS and asterisk). Since that time, channel 1 on my ZAP card has been unuseable. I get a "device already in use" message (I need to restart after-hours to catch the exact error message), and asterisk won't start. This hasn't been a real big deal, as we have 3 other analog extensions available. I just noticed this morning a difference between 2 systems though that may be the issue, but not sure why: ON the asterisk system that is NOT working, `zap show channels` gives me: asterisk-bvr*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo intern-hcst default 2 intern-hcst default 3 intern-hcst default 4 intern-hcst default 6 inbound default 7 inbound default 8 inbound default While on the one that DOES work, I see: asterisk-home*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo from-pstn en 1 from-pstn en 4 from-internal en The difference being (that I see) is that on asterisk-home, the "pseduo" channel is on a separate channel, where as on the problem system, it is (I think) sitting on channel 1. I also notice that if I unload chan_zap.so, it will say it is unregistering channel 1, which leads me to believe that SOMETHING is causing it to occupy that zap channel. Are there any clues here as to why this is? Barry D. Hassler President HCST 2332 Grange Hall Road Beavercreek, Ohio 45431-2345 http://www.hcst.net/ [EMAIL PROTECTED] +1 937-427-9000 +1 937-427-8706 FAX FWD: 3934279000 (655480) HCST*Net Support Issues: please email [EMAIL PROTECTED] Billing Issues: Please email [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Grandstream GX-2000 phones for paging
Firmware 1.1.1.9? Where's that? most recent on Grandstream's site that I see is 1.1.0.16, which I have loaded.
On Fri, 2006-09-08 at 20:20 -0500, Lacy Moore - Aspendora wrote:
You also have to make sure that on the web config for Grandstream that you allow it to receive auto-answer (or something to that effect).
Ok, actually it's under the settings for the Lines and is called: Allow Auto Answer by Call-Info:
Make sure Yes is selected here.
You can use what Barry has below for paging (or rather intercom) to a single phone. For actual paging (i.e., several phones), use the Page command (show application page for options from the CLI). On paging, I would recommend this: Turn off speaker on remote disconnect: be set to Yes as well.
This works fine for me on firmware 1.1.1.9.
On 9/8/06, Barry D. Hassler <[EMAIL PROTECTED]> wrote:
This isn't working for me either. I was about to ask this same question, but discovered this recent thread.
I have the following set up in my extensions.conf file, as per Granstream instructions:
[macro-page-grandstream]
exten => s,1,ChanIsAvail(${ARG1}|js); j is for jump, s is for ANY call
exten => s,2,SIPAddHeader(Call-Info: answer-after=0)
exten -> s,3,Dial(${ARG1})
exten => s,4,NoOp();
exten => s,5,Hangup
exten => s,102,NoOp(102) ; Channel not available
exten => s,103,Hangup
[intercoms]
exten => **2311,1,Macro(page-grandstream,SIP/2311)
exten => **2311,2,Hangup
And in my local context:
include => intercoms
When I dial **2311, I see the following debug output:
[Sep 8 15:24:37] -- Starting simple switch on 'Zap/4-1'
[Sep 8 15:24:43] -- Executing SetMusicOnHold("Zap/4-1", "default") in new stack
[Sep 8 15:24:43] -- Executing Goto("Zap/4-1", "intern-hcst-post|**2311|1") in new stack
[Sep 8 15:24:43] -- Goto (intern-hcst-post,**2311,1)
[Sep 8 15:24:43] -- Executing Macro("Zap/4-1", "page-grandstream|SIP/2311") in new stack
[Sep 8 15:24:43] -- Executing ChanIsAvail("Zap/4-1", "SIP/2311|js") in new stack
[Sep 8 15:24:43] -- Executing SIPAddHeader("Zap/4-1", "Call-Info: answer-after=0") in new stack
[Sep 8 15:24:43] -- Executing Hangup("Zap/4-1", "") in new stack
[Sep 8 15:24:43] == Spawn extension (intern-hcst-post, **2311, 2) exited non-zero on 'Zap/4-1'
[Sep 8 15:24:43] -- Hungup 'Zap/4-1'
Is this a problem with the SIPAddHEader that it is jumping immediately to Hangup? I see NO SIP traffic as a result of this, and sip debug shows nothing out of the ordinary.
The BLF functions don't seem to be working either.
I'm running asterisk 1.2.9.1, and have the Granstream GXP2000 reports:
Software Version: Program-- 1.1.0.16 Bootloader-- 1.1.0.1
On Sat, 2006-09-02 at 20:31 -0500, Larry Alkoff wrote:
Nic Bellamy wrote:
> Zeeshan Zakaria wrote:
>
>> My client has all Grandstream GX-2000 phones in his office and he
>> wants receptionist to use them for paging as well. Currently they are
>> using Nortel and receptionist can easily do paging. He said that he
>> had somebody setup their old Asterisk system in a way, that
>> receptionist could dial an extension, after which her voice was heard
>> on all grandstream phones' speaker phones.
>>
>> I want to know how to setup this type of feature on grandstream
>> phones, i.e. dialing an extension will activate all phones' speaker
>> phones.
>
> http://www.grandstream.com/FAQ/Asterisk.htm
>
> There's a PDF there that tells you (a) what settings to put on the
> phone, and (b) how to configure Asterisk to sent the SIP header that
> tells the phone to auto-answer.
>
> Cheers,
>Nic.
>
Please let me know if you get this working. I couldn't.
Larry
Barry D. Hassler
President
HCST
2332 Grange Hall Road
Beavercreek, Ohio 45431-2345
http://www.hcst.net/
[EMAIL PROTECTED]
+1 937-427-9000
+1 937-427-8706 FAX
FWD: 3934279000 (655480)
HCST*Net Support Issues: please email [EMAIL PROTECTED]
Billing Issues: Please email [EMAIL PROTECTED]
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Re: [asterisk-users] How to use Grandstream GX-2000 phones for paging
This isn't working for me either. I was about to ask this same question, but discovered this recent thread.
I have the following set up in my extensions.conf file, as per Granstream instructions:
[macro-page-grandstream]
exten => s,1,ChanIsAvail(${ARG1}|js); j is for jump, s is for ANY call
exten => s,2,SIPAddHeader(Call-Info: answer-after=0)
exten -> s,3,Dial(${ARG1})
exten => s,4,NoOp();
exten => s,5,Hangup
exten => s,102,NoOp(102) ; Channel not available
exten => s,103,Hangup
[intercoms]
exten => **2311,1,Macro(page-grandstream,SIP/2311)
exten => **2311,2,Hangup
And in my local context:
include => intercoms
When I dial **2311, I see the following debug output:
[Sep 8 15:24:37] -- Starting simple switch on 'Zap/4-1'
[Sep 8 15:24:43] -- Executing SetMusicOnHold("Zap/4-1", "default") in new stack
[Sep 8 15:24:43] -- Executing Goto("Zap/4-1", "intern-hcst-post|**2311|1") in new stack
[Sep 8 15:24:43] -- Goto (intern-hcst-post,**2311,1)
[Sep 8 15:24:43] -- Executing Macro("Zap/4-1", "page-grandstream|SIP/2311") in new stack
[Sep 8 15:24:43] -- Executing ChanIsAvail("Zap/4-1", "SIP/2311|js") in new stack
[Sep 8 15:24:43] -- Executing SIPAddHeader("Zap/4-1", "Call-Info: answer-after=0") in new stack
[Sep 8 15:24:43] -- Executing Hangup("Zap/4-1", "") in new stack
[Sep 8 15:24:43] == Spawn extension (intern-hcst-post, **2311, 2) exited non-zero on 'Zap/4-1'
[Sep 8 15:24:43] -- Hungup 'Zap/4-1'
Is this a problem with the SIPAddHEader that it is jumping immediately to Hangup? I see NO SIP traffic as a result of this, and sip debug shows nothing out of the ordinary.
The BLF functions don't seem to be working either.
I'm running asterisk 1.2.9.1, and have the Granstream GXP2000 reports:
Software Version: Program-- 1.1.0.16 Bootloader-- 1.1.0.1
On Sat, 2006-09-02 at 20:31 -0500, Larry Alkoff wrote:
Nic Bellamy wrote:
> Zeeshan Zakaria wrote:
>
>> My client has all Grandstream GX-2000 phones in his office and he
>> wants receptionist to use them for paging as well. Currently they are
>> using Nortel and receptionist can easily do paging. He said that he
>> had somebody setup their old Asterisk system in a way, that
>> receptionist could dial an extension, after which her voice was heard
>> on all grandstream phones' speaker phones.
>>
>> I want to know how to setup this type of feature on grandstream
>> phones, i.e. dialing an extension will activate all phones' speaker
>> phones.
>
> http://www.grandstream.com/FAQ/Asterisk.htm
>
> There's a PDF there that tells you (a) what settings to put on the
> phone, and (b) how to configure Asterisk to sent the SIP header that
> tells the phone to auto-answer.
>
> Cheers,
>Nic.
>
Please let me know if you get this working. I couldn't.
Larry
Barry D. Hassler
President
HCST
2332 Grange Hall Road
Beavercreek, Ohio 45431-2345
http://www.hcst.net/
[EMAIL PROTECTED]
+1 937-427-9000
+1 937-427-8706 FAX
FWD: 3934279000 (655480)
HCST*Net Support Issues: please email [EMAIL PROTECTED]
Billing Issues: Please email [EMAIL PROTECTED]
___
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Re: [asterisk-users] Linksys SPA-3102
On Sat, 2006-08-19 at 00:12 -0500, Rich Adamson wrote: Barry D. Hassler wrote: > Any further experience with the 3102? I'm looking for a solution to > connect 2 CO lines and a set of 2-line phones to my asterisk server > (along with a bunch of SIP phones). Would 2 of these work well for that? > > Hopefully no echo problems! That would kill this project? I'm still > searching for one peice of hardware that would have 2 FXO and 2 FXO > ports on it, but haven't stumbled on it yet The only way to know for sure whether the spa will provide reasonable service is to try it on the actual pstn line to be used. There is no other way for anyone to tell you anything different. The quality of the audio & echo is 100% dependent on the exact pstn line characteristics, etc. Two other reasonable alternatives are: Digium TDM card with two fxo and two fxs modules. Sangoma A200d card with one fxo module (has two lines on it) and one fxs module (has two lines on it). Yes, this would definitely work. J'm trying to avoid running phone lines the distance between the existing demarc and the asterisk server :-) But, it would probably be the best solution. Actually, as I look further, I don't think the 3102 will work in this environment at all for what I want. (incoming FXO calls go to Asterisk server, and through its dial plan, which might include ringing the analog phones on the FXS ports). I did find AudioCodes MediaPack MP-114, which has exactly the configuration I want (2 FXO, 2 FXS), but have to be registered to get any documentation on the units. 2 Grandstream HT-488's looks like a possibility too I think I'll just run the phone lines into the server, and add more cards there :-) The only way to know for sure whether the TDM card will provide reasonable service is to try it on the actual pstn lines, just exactly like the spa box. Same issues as the spa; some pstn lines work fine, others have echo that nags users. The Sangoma card with h/w echo canceler just plain works, but is probably the more expensive of the alternatives. The Mediatrix 1204 seems to have excellent echo characteristics, but it is likely the most expensive approach for small systems. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA-3102
Any further experience with the 3102? I'm looking for a solution to connect 2 CO lines and a set of 2-line phones to my asterisk server (along with a bunch of SIP phones). Would 2 of these work well for that? Hopefully no echo problems! That would kill this project? I'm still searching for one peice of hardware that would have 2 FXO and 2 FXO ports on it, but haven't stumbled on it yet On Thu, 2006-07-27 at 09:33 -0400, Wes Baehr wrote: Has anyone used the new 3102? If so, does it work correctly? I heard lots of horror stories about the SPA-3000 causing terrible echo, picking up voice tones as DTMF, etc, so I’m a little hesitant to buy. Thanks, Wes Baehr -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/401 - Release Date: 7/26/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI vs "Digital Trunk"
Thanks for the answers folks. I think I now understand that indeed, the "Digital Trunk" is really nothing more than the same analog services, just delivered digitally. no outbound ANI (of especial interest), etc This is in the Dayton Ohio area, major telco vendor. I'll have to pursue the differences with them further, as the pricing is about double for the PRI vs the "Digital Trunk". I'd like to move 7 analog lines to a digital interface, but just can't cost-justify it in this scenario :-( On Tue, 2006-07-25 at 15:25 -0400, Barry D. Hassler wrote: Hi, can someone enlighten me as to the difference between a PRI and a "Digital Trunk" (other than cost)? I do understand PRI (B-channel signaling, incoming/outgoing call setup, D channel for voice/data, etc), but I'm not quite sure how that compares with what my vendor is calling a "Digital Trunk" (specifically in contrast to a PRI). The PRI is about twice the cost. If this is just a channelized T1 (24 64k voice/data "channels'), would they each be assigned a specific phone number, or is there further flexibility in sending/receiving calls, callerid (receive or send), etc? Feeling ignorant here 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 Barry D. Hassler President HCST 2332 Grange Hall Road Beavercreek, Ohio 45431-2345 http://www.hcst.net/ [EMAIL PROTECTED] +1 937-427-9000 +1 937-427-8706 FAX FWD: 3934279000 (655480) HCST*Net Support Issues: please email [EMAIL PROTECTED] Billing Issues: Please email [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI vs "Digital Trunk"
Hi, can someone enlighten me as to the difference between a PRI and a "Digital Trunk" (other than cost)? I do understand PRI (B-channel signaling, incoming/outgoing call setup, D channel for voice/data, etc), but I'm not quite sure how that compares with what my vendor is calling a "Digital Trunk" (specifically in contrast to a PRI). The PRI is about twice the cost. If this is just a channelized T1 (24 64k voice/data "channels'), would they each be assigned a specific phone number, or is there further flexibility in sending/receiving calls, callerid (receive or send), etc? Feeling ignorant here 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
