[asterisk-users] DAHDI, PRI and callerid
Hi, I have just set up Asterisk to use an E1 line with a Digium card. And I can call both in and out, but my outgoing line is all ways identifying itself as the same number, and i can't even change it to another number in the same number series. Do anyone have some clue on how to fix this. I'm using Asterisk 1.6.2.13, libpri 1.4.11.4 and DAHDI 2.4.0. /etc/dahdi/system.conf: span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-12 unused=13-15,17-31 dchan=16 echocanceller=mg2,1-12 # Global data loadzone= no defaultzone = no /etc/asterisk/chan_dahdi.conf [trunkgroups] trunkgroup = 1,16,31 spanmap = 1,1,1 [channels] context=incoming usecallerid=yes callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes usecallerid=yes callerid=asreceived usecallingpres=yes callgroup=1 switchtype = euroisdn signalling = pri_cpe group = 1 channel = 1-12 Best regards, Kent Varmedal -- _ -- 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
[asterisk-users] How to use Atxfer in AMI
Hi, I'm trying to make a attended transfer through AMI. I though i could use Atxfer, and it seems ok, but nothing happens. And I can't find any how-to or description on how to do this. What more do I have to do to make this work? In Asterisk Call Manager: Action: Atxfer Channel: SIP/36-xx Exten: 33 Priority: 1 Context: Phone Response: Success Message: Atxfer successfully queued Best regards, Kent Varmedal -- _ -- 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
[asterisk-users] How to set up Asterisk to deliver a trunk sip connection?
Hi, I'm trying to set up an old PBX (that supports SIP) to go through our new Asterisk server, so that our old phones can be used still for some time. How can I set up Asterisk to deliver a trunk sip connection that our old PBX can connect to? Is it just to sett up a normal sip device in sip.conf? Or is there some other / extra magic for this to work? I can't test this with the old system before we go live with Asterisk (and then it must work). But I can't seem to find any concrete information about this. Best regards, Kent Varmedal -- _ -- 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] How to set up Asterisk to deliver a trunk sip connection?
On Wed, 2010-08-11 at 10:29 -0400, David Backeberg wrote: On Wed, Aug 11, 2010 at 10:12 AM, Kent Varmedal k...@domeneshop.no wrote: I'm trying to set up an old PBX (that supports SIP) to go through our new Asterisk server, so that our old phones can be used still for some time. How can I set up Asterisk to deliver a trunk sip connection that our old PBX can connect to? Is it just to sett up a normal sip device in sip.conf? Or is there some other / extra magic for this to work? Pretty much. The details vary on codec / DTMF, etc. based on what you're talking to, but that's the general idea. I can't test this with the old system before we go live with Asterisk (and then it must work). Really? Why not? If it speaks SIP, it should be able to do multiple SIP trunks / channels, and you should be able to set up a simultaneous SIP trunk alongside your production line(s). We need to upgrade this PBX for it to work with SIP, it is at the moment using ISDN. And those who delivered it and do the support/reconfiguration is paid by the hour. We don't have any control over it our self, so when it is changed it will stay that way. If you tell us the PBX, somebody here has probably worked with it. It is a Ascotel intelliGate 2025/2045. If the old PBX speaks true SIP, you could ditch the old PBX, and have the SIP phones register directly with asterisk. We don't have any SIP phones at the moment, we use DECT phones from Aastra (not the IP version). New phones will probably be SIP. Best regards, Kent Varmedal -- _ -- 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] OT: Opensource Sipura Profile Compiler for SPA2K, 3K
On Fri, 01 Oct 2004 23:16:45 -0500, Kristian Kielhofner [EMAIL PROTECTED] wrote: Kristian Kielhofner wrote: They do, but the format of that file has to be generated with Sipura's proprietary config tool. Currently NOT available for anyone that Sipura doesn't want to have it. Actually part of that is not true. The Sipura's will take a txt file that has an XML like format, this is documented by Sipura. You can tell it where to get the file in the provisioning rules or have it call a cgi script that generates the file on the fly. The file can even be encrypted with a standard AES 256bit cipher key that you have pre-shared with the device. You do not need the Sipura tool to generate these files. You can build a simple php or perl app to generate the file since its just text with XML like tags. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Opensource Sipura Profile Compiler for SPA2K, 3K
On Fri, 22 Oct 2004 14:12:19 -0500, Kristian Kielhofner [EMAIL PROTECTED] wrote: Kent, Where is this documented? There is a provisioning document available from Sipura. Now that I have looked on their web site that document does not appear to be available unless you have a support login. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mwi over serial port
We have the SMDI interface running to a DMS 10. If anyone is interested let us know. The code would need a little clean up to get released. Kent On 10/13/04 10:29 AM, Michael Welter [EMAIL PROTECTED] wrote: The bounty is bogus, the offerors are not serious, and they should take it off the wiki. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk install in a home with regular phones and a x100p
I use the following extension in my dial plan to pickup the ZAP channel but not dial anything. This works very well and allows me to pickup a call from a VOIP phone after the call has already been answered by someone else in the house on the POTS line. exten = ,1,Dial,ZAP/1 ; Pickup analog line but don't dial anything On Mon, 20 Sep 2004 20:09:26 -0400, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello all, I have a question. I have an asterisk box setup with an x100p card installed. I have about 3 VoIP phones connected to it. The x100p is connected to my pots line. I also have regular phones connected the the pots line as well like most normal people would. When I get an incomming call all the regular phones ring and my VoIP phones ring. This is good. My question is this. Say a call comes in on the pots line and a someone answers it on a regular phone and the call was for me and I am by a VoIP phone. Is there anyway to get asterisk to pickup the X100P card and route the call to the VoIP phone i am on just like as if you pick up another regular phone in the house. Setting up an extention to do this would be nice. I looked at the call grouping stuff but that only seems to be for if the phone is currently ringing. Thanks for any help -Lee This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pickup zap channel already in use?
I'm getting a ton of noise on the channel just from the * side when I pickup my zap channel. Otherwise it works fine, if the other person in the house hangs up the noise goes away.. On Tue, 27 Jul 2004 23:30:08 -0400, Mark Woods [EMAIL PROTECTED] wrote: Kent wrote: On Tue, 27 Jul 2004 16:58:47 +, Mark Woods [EMAIL PROTECTED] wrote: I have to admit that your question interested me because I'm thinking of setting up the same thing. As of yet, though, I haven't found an answer to it. It's fairly simple when * has picked up, but I haven't really devoted much time to figuring out how to do it when it hasn't. So...let me work on it, and I'll let you know what I come up with. It's going to take at least a week, though, as I'm going to Oshkosh for the EAA Airventure on Thursday...but I'll see what I can come up with after that. -Mark A friend of mine who is another * user suggested using an extension with an empty Dial statement to connect my sip phone to the zap channel. I am going to try that tonight and see if that works. Let me know if you figure out anything else. Thanks! I actually got a chance to try it just now. Works like a champ! It has the added benefit of giving direct access, with dialtone, to the outside line, instead of having * dial. Here's what I put in my extensions.conf: exten = 4000,1,Dial(Zap/1/) exten = 4000,2,Congestion -Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pickup zap channel already in use?
No, Asterisk saw the call but someone has already picked it up via the telco line NOT via Asterisk. So * shows as no call connected and Zap channel not in use. On Tue, 27 Jul 2004 0:23:26 +, Mark Woods [EMAIL PROTECTED] wrote: In your scenario, has asterisk picked up at all? -Mark I am using asterisk at home with a Cisco ATA186 and a clone X100P card. My inbound telco line is plugged into the X100P card. My telco line is also plugged into other phones in the house for now so someone else can answer the phone without asterisk being involved. What I would like to do is if someone has answered the call on a normal phone in the house I would like to be able to join the call from a SIP phone by dialing an extension or feature code. Is there any way to do this? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Pickup zap channel already in use?
On Tue, 27 Jul 2004 16:58:47 +, Mark Woods [EMAIL PROTECTED] wrote: I have to admit that your question interested me because I'm thinking of setting up the same thing. As of yet, though, I haven't found an answer to it. It's fairly simple when * has picked up, but I haven't really devoted much time to figuring out how to do it when it hasn't. So...let me work on it, and I'll let you know what I come up with. It's going to take at least a week, though, as I'm going to Oshkosh for the EAA Airventure on Thursday...but I'll see what I can come up with after that. -Mark A friend of mine who is another * user suggested using an extension with an empty Dial statement to connect my sip phone to the zap channel. I am going to try that tonight and see if that works. Let me know if you figure out anything else. Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pickup zap channel already in use?
I am using asterisk at home with a Cisco ATA186 and a clone X100P card. My inbound telco line is plugged into the X100P card. My telco line is also plugged into other phones in the house for now so someone else can answer the phone without asterisk being involved. What I would like to do is if someone has answered the call on a normal phone in the house I would like to be able to join the call from a SIP phone by dialing an extension or feature code. Is there any way to do this? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zhone Zplex issues
Hey guys, I've had a working Asterisk setup going for a while now, but am having problems with the Zhone Zplex 10b thinking that during ringing, an extension has answered the call when infact it hasn't. This only seems to happen on some of the ports and doesn't appear to be specific to the handset. Does anyone have any suggestions as to how I could go about fixing this (aside from throwing the Zhone out)? Cheers, -Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zhone + call transfer
After finding a spot to put the Zhone Zplex so that the fan noise doesn't annoy anyone, I've got everything working to an acceptable level except for call transfers. No matter what I do the Zhone doesn't seem to be passing 'flash' key presses on to asterisk, ie whenever I try to transfer a call, nothing happens. The DTMF tones pressed after the 'flash' key are simply heard over the conversation. Running asterisk with -vvvc doesn't show anything when trying to transfer a call which leads me to believe that it has something to do with the Zhone. Can anyone confirm that call transfers do in fact work with the Zhone Zplex? Is there anything obvious that I may have missed? ...and yes, I've added the following to Zapata.conf for the appropriate channels: threewaycalling = yes transfer = yes cancallforward = yes Cheers, Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WTB / WTS Voip hardware
-Original Message- From: Ken Alker [mailto:[EMAIL PROTECTED] Sent: Monday, 12 January 2004 5:47 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] WTB / WTS Voip hardware --On Sunday, January 11, 2004 5:24 PM +1100 Kent Williams [EMAIL PROTECTED] wrote: I've got a Wildcard T100P along with a Zhone Zplex 10 24S/O which has been working fine for me now for a while. These have been pulled out of a working Asterisk installation (as they were no longer required) Why'd you tank Asterisk from your work place? Did it not perform for you? No real reason in particular, just the callerId issue and weren't willing to shell out for something like the Adtran offerings. to use at home only to find that the fan noise is too loud. I assume the only thing with loud fan noise is the Zhone Zplex, right? Yeah, no matter where I put it, I can always hear it :) As such I'm looking to sell off this hardware and replace it with some combination of fanless hardware that will allow me to have 4 handsets and 2 incoming lines. I don't really want to spend any more than the money I'd make from selling the T100P / Zplex. So, is anyone interested in buying this gear / selling some of their old gear / trading? I *might* be interested. We just built our first Asterisk server at work and I'm trying to decide what hardware to buy. I'm seriously considering a T100P with some type of channel bank. I've never heard of or seen the Zplex but I found it buried on Zphone's web site. They don't list in in their product list, however. Is is discontinued? I also noticed on past postings that perhaps the Zplex 10 doesn't support FXO CallerID. Do you know if that was resolved? The documentation at http://www.zhone.com/support/technical_support/zplex/product_documents/ ug _ 0011_10b.pdf claims it can do FXO CallerID. What is the difference between the 10A and 10B, and which do you have? From memory, the 10A is FXS only. The 10B has 8 ports that can be either FXS or FXO. I've got the 10B Does the Zplex have echo issues? I haven't had any serious issues, not sure about other people. Any other suggestions? Cheers, Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users /** Ken Alker [EMAIL PROTECTED]ham radio: KA6SDU Impulse Internet Services http://www.impulse.net Santa Barbara, San Luis Obispo, Ventura, Los Angeles, Orange T-3 / T-1 / ADSL / ISDN / 56K / web hosting / wireless / co-lo ***/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WTB / WTS Voip hardware
I've got a Wildcard T100P along with a Zhone Zplex 10 24S/O which has been working fine for me now for a while. These have been pulled out of a working Asterisk installation (as they were no longer required) to use at home only to find that the fan noise is too loud. As such I'm looking to sell off this hardware and replace it with some combination of fanless hardware that will allow me to have 4 handsets and 2 incoming lines. I don't really want to spend any more than the money I'd make from selling the T100P / Zplex. So, is anyone interested in buying this gear / selling some of their old gear / trading? Any other suggestions? Cheers, Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to flash hook when there is no hook ?
I just looked at my cordless phone and it has a flash button. It's a Panasonic... Remi Letot wrote: Hi all, I'm using regular analog phones with * and an ATA186, and I plan to move to cordless phones. But cordless phones usually have no way to flash hook cause there is no hook :-) I seem to recall there is a way to simulate a flash hook using dtmf tones in *, but I can't find which combination to use. More generally, where can I find all those magic cominations to put people on hold, use pickup grouping, redirect calls,... ? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone using * in a live production environment?
Steven Critchfield wrote: On Tue, 2003-11-04 at 03:48, Gavin Hamill wrote: Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case study, just a few lines to try and get a grip on the number of real installations. I'm not trying to flame you for this message, but this is something that is asked about quite often and doesn't exactly prove anything. Just because I successfully pulled off an installation doesn't mean you will be successful. There is obviously working systems out here otherwise there wouldn't be this much traffic on this list. Maybe what needs to happen to keep this question from coming up over and over again would be to get this on the wiki. Someone want to create the page and then post the link to that specific page to be filled in? Actually, this is the thing I find most usefull. There are not very many posts where the statement is about a fully functional asterisk system being deployed in a business. The vast majority of posts are about problems with asterisk, and in my case it has kept me from replacing our functional but aging PBX with asterisk. More success stories with details about the hardware and the environment would be of great benefit to this list, in my opinion. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zhone Zplex 10 units
Mine has been working well, but the only problem is that it doesn't support callerid (from the POTS side). -Original Message- From: John Schmerold [mailto:[EMAIL PROTECTED] Sent: Tuesday, 5 August 2003 12:37 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zhone Zplex 10 units Thanks for the Zplex heads up. Steven Critchfield wrote: On Mon, 2003-08-04 at 13:13, John Schmerold wrote: I frequently see Zhone Zplex 10 units on Ebay - cheap. What's the story on these? Are they flaky? search the archives. Tough to configure? tough, no, pain in the ***, yes Any other issues that come to mind? search the archive, that is why it is there. I don't see them listed on Zhone's website (except in support), so I suspect they've discontinued the product, but if it's a good product I could use it to learn Asterisk. Thats funny since they don't really even act like they want to support them. -- John Schmerold Katy Computer Systems, Inc 20 Meramec Station Rd Valley Park MO 63088 314-316-9000 v 775-227-6947 f ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Products for use in Australia
Hey guys, I've noticed a bit of traffic lately about products (or lack of) that can be used in Australia legally. At the moment I have a zhone zplex and t100p running with asterisk but am looking to change for the following reasons: 1. No incoming callerid on the zhone 2. Fan noise of the zhone 3. Not being compliant with the stupid Aussie telco regulations I want to sell the t100p and zhone to replace it with a setup that allows me to have 2 incoming lines from the telco and 4 handsets. I seem to remember a pci card being mentioned that had 6 ports, each configurable for fxo or fxs (although I could be imagining things as I couldn't find the relevant post in the archives). Anyways, Is anyone aware of any products that will suit my application? Is there any plan to certify the digum products for use in Australia? Are there any products about that would be fine, but aren't supported by Asterisk? Is there anyone working on a product with 6 ports, configurable as FXO/FXS? Cheers, Kent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
