Re: [asterisk-users] Asterisk 1.4 and chan_misdn
Hi, you also need mISDNuser. After that make clean make install you'll have access to chan_misdn. Regards. Pierre Administrator TOOTAI wrote: Hi list, I installed a fresh Debian/Etch with Asterisk 1.4 and Zaptel 1.4 from SVN for 2 Digium B410P card. I ran configure in Asterisk dir, went in zaptel dir and: make, make install, make b410p. Everything is ok. Now I want to compile Asterisk but can't activate the chan_misdn channel which depends on -from menuselect- isdnnet(E), misdn(E), suppserv(E) When I made the make b410p, all the misdn stuff was downloaded from digium's ftp. Also, running /etc/init.d/misdn-init --scan show me the 2 cards I have, /etc/init.d/misdn-init --config prepare me the misdn.conf and after a /etc/init.d/misdn-init start I see: mISDN_dsp 191656 0 mISDN_capi 88716 0 mISDN_l2 34452 0 mISDN_l1 11036 0 mISDN_core 71360 6 mISDN_dsp,hfcmulti,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1 kernelcapi 44576 2 mISDN_capi,capi My questions: why Asterisk doesn't want to let me activate the misdn channel? Is misdn ready for 1.4? Thanks for any hint ___ --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] Moving from Bristuff to mISDN
On 26 Mar 2007, at 12:34, Olivier wrote: Hi, Beside having to use misdn.conf instead of zaptel.conf, did you notice any gain or lost moving from bristuff to misdn ? I was thinking about callerID, compliance to Telco ISDN, ... We have had reports that misdn causes asterisk to emit 128byte alaw packets instead of 160bytes. Many endpoints seem to be ok with this, but corraleta for one doesn't like this one little bit :-( Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Multi-registration ?
On 26 Mar 2007, at 16:33, Olivier wrote: Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? While this is possible it isn't easy, You will trip over resource sharing problems in 2 areas: 1) port numbers - SIP likes to listen on 5060UDP - you could configure each softphone to use a non-standard port, but you have to tell asterisk that, and any other SIP aware network infrastructure (NAT routers etc) 2) Audio hardware - assuming you only have one speaker and microphone on your PC, which softphone should get the audio data? How will they decide? 2. Is possible to do the same with SIP hardphones ? Some hardphones support registering to multiple sip accounts from one phone. (as indeed do some softphones) Is that what you want ? Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and T38 ?
Noc Phibee schrieb: Hi i read the list and see a lot of personn say T38 it's not possible with asterisk and other says that he use T38 with asterisk ?? i don't understand ;=) Well, if i understand it correctly then Asterisk currently only supports T.38-Passthrough, which means, you have to have to T.38 capable Endpoints which can communicate with an Asterisk in the Media Path. But you cannot terminate an T.38 Call on an Asterisk Server (say receiving an Fax with an Asterisk and saving the Fax as an TIFF on the server). Anyone feel free to correct me if i am wrong ;) Cheers, Tobias ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and T38 ?
Tobias Wolf a écrit : Noc Phibee schrieb: Hi i read the list and see a lot of personn say T38 it's not possible with asterisk and other says that he use T38 with asterisk ?? i don't understand ;=) Well, if i understand it correctly then Asterisk currently only supports T.38-Passthrough, which means, you have to have to T.38 capable Endpoints which can communicate with an Asterisk in the Media Path. But you cannot terminate an T.38 Call on an Asterisk Server (say receiving an Fax with an Asterisk and saving the Fax as an TIFF on the server). Anyone feel free to correct me if i am wrong ;) Cheers, Tobias Thanks for your answer ;=) We don't have a solution for use a codec without comrpession for supply a line at a Fax and at a modem ? Modem/Fax with VoIP never work ? ___ --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] Limit call duration
Yes you can use the L flag but i dont know if there is any system variable used for this purpose. On 3/26/07, Suity Zsolt [EMAIL PROTECTED] wrote: Rizwan Hisham wrote: I think you can set absolute timeout variable for incoming call also. I havent tested it yet, y dont you try it. do like this: before every local extension you can set: exten= _XXX,1,SET(Timeout(absolute) = 10) exten= 123,2,Dial exten= 234,2,Dial Yuan Liu I think you can say something like: AbsoluteTimeout (or in 1.2x, Set(TIMEOUT(absolute) = seconds) ) Thank you! It works as I expected. Can I set some system var to warn caller how many time is left? (like in Dial L flag) -- Suich ___ --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 -- Regards Rizwan Hisham Software Engineer ___ --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] SRTP vs ZRTP in Asterisk
On 26 Mar 2007, at 22:32, Michael Graves wrote: Hi All, I've been reading about Phil Zimmermann's ZRTP encryption scheme for SIP clients. This seems attactive but I don't use soft phones. I'm guessing that we'd need ZRTP support in Asterisk in order to use it to secure calls from hard phones. There seem to be issues with SRTP key exhange between various devices. So much so that the IETF is working on a standardization project. ZRTP, which is one of the proposals before the IETF, overcomes this. Since Zimmermann has open sourced the protocol I would hope that it could be implemented in Asterisk without too much trouble. Does the current work on SRTP extend into ZRTP? At Etel I heard Phil Zimmermann say that he had a working implementation of ZRTP for asterisk in the lab. What was less clear was how/when this might be released. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] vzaphfc installation...
Hi everybody, does anybody knows how to install and configure VZAPHFC? Thank you Best regards Mauro ___ --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] ztdummy and MOH
Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI zap show status Description Alarms IRQbpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct. Please help. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: how to define a pilot number
is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Lito I'm assuming you are talking about analog lines as PRI's will do this more-or-less naturally. This is a telco feature as opposed to an Asterisk feature. Here in Bell Canada country they call it Ringer Equivalence. Call your local carrier and they should be able to tell you what they call it in their marketing world. You tell the telco which lines you want the calls to roll to then all three will terminate calls to the pilot number. Now it doesn't work exactly as you had described - it doesn't move the call so as to free up the first port. It merely says the first port is busy and terminates the next call on the next port in sequence. This means you can't count on which line is available at any time. For outbound, you need to put the three lines in an Asterisk group and test the group for availability to select an available line to dial out on. dbc. -- David Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using server side phonebook directory with SPA941
Hello list, I got a couple of those wouldn't it be great questions, as following: 1. Is it possible, with asterisk to hold a central phonebook directory of callers?, so that when this party calls a textual caller ID will be displayed on the phone display. 2. How can this be configured with Trixbox, I've looked at the configuration options - I assume it plays no difference me basing it on mysql or astdb? 3. What protocol does the phone (Linksys SPA941) talks to the asterisk server to retrieve this information ? 4. Has someone done this? What softphone should I use to test it first (I'm connecting it with outlook, so it has to be win* software) Thanks for helping, Maxim. -- Cheers, Maxim Veksler Free as in Freedom - Do u GNU ? ___ --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] AMI - delete voicemail
How can I delete voice mails (all, new and old) from AMI? I thought that I could use Action Command, but there is no command to delete voicemail. So I figure it out to use system command and execute rm /var/spool/asterisk/voicemail/default/100/INBOX/* and rm /var/spool/asterisk/voicemail/default/100/tmp* but I don't know how to do that from AMI. Any suggestions how to do this are welcome. -- Tomislav Parcina [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] snom led not working with asterisk 1.4.1
Steve Murphy wrote: On Wed, 2007-03-21 at 15:00 +0100, Giorgio Incantalupo wrote: Hi Steve, as you know if you type show hints inside asterisk console you can see phone status. When a phone is not connected, Asterisk says it is Unavailable. With Asterisk 1.2.9.1 my SNOM leds worked well so I knew when a phone was not available but with Asterisk 1.4.1 is not possible anymore. This is one of the functions which I'm trying to keep from Asterisk 1.2.9.1 to 1.4.1 . Pardon my ignorance! I am new in this area. I have not used my SNOM 360 with anything but 1.4. When the monitored extension is busy, the LED is on; when the extension is ringing, the LED flashes. What does it do for you in 1.2, when the line is unavailable? The LED is also on. I noticed a change from 1.2 to 1.4: channels/chan_sip.c, 1.2.13 : case AST_EXTENSION_UNAVAILABLE: statestring = confirmed; local_state = NOTIFY_CLOSED; pidfstate = away; pidfnote = Unavailable; Asterisk 1.4.2 channels/chan_sip.c Line 6892 function static int transmit_state_notify(...) case AST_EXTENSION_UNAVAILABLE: statestring = terminated; local_state = NOTIFY_CLOSED; pidfstate = away; pidfnote = Unavailable; The var statestring has changed. I changed it back to confirmed and the phone shows the unavailable state. ciao, Carsten ___ --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: Question about DSP in Digium card
well, ...,we did not choose SIP because our customers are located behind NAT router (using private IP's) and those routers are not managed by them but by the ISP so it is very difficult to establish full duplex phone calls because you can not initiate voice over ip session from the internet (outside) to LAN side (inside) with private IP's. We could not establish 2-way phone calls, I mean, the conversation is listened in 1-way only. As I mentioned before, we can not configure PAT into the NAT router neither because is handled by the ISP and the passwords are unknown That's why we decided to use IAX instead of SIP, I mean, IAX is more robust than SIP when the NAT router is 3th-party managed and the PAT feature is not enable. On the other and we tested IAX over dialup links and it worked fine Those are the reasons we choose IAX as acess protocol to our SIP/H323 Network. You know, the access networks of the customers are different completely: Private IP Address over DSL lines (NAT Router), Public IP Address over DSL lines, Corporate Networks over dedicated Links (Public and IP Addresses), Dialup links, .. Any comment would be welcomed, thanks a lot Levy.- 2007/3/24, A. Levy [EMAIL PROTECTED]: Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX - ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and T38 ?
Just for my 2cents.. Faxing can and does work over G711u. We do it. Although it stops working when your Internet connection gets jittery, it does work. On 3/27/07, Noc Phibee [EMAIL PROTECTED] wrote: Tobias Wolf a écrit : Noc Phibee schrieb: Hi i read the list and see a lot of personn say T38 it's not possible with asterisk and other says that he use T38 with asterisk ?? i don't understand ;=) Well, if i understand it correctly then Asterisk currently only supports T.38-Passthrough, which means, you have to have to T.38 capable Endpoints which can communicate with an Asterisk in the Media Path. But you cannot terminate an T.38 Call on an Asterisk Server (say receiving an Fax with an Asterisk and saving the Fax as an TIFF on the server). Anyone feel free to correct me if i am wrong ;) Cheers, Tobias Thanks for your answer ;=) We don't have a solution for use a codec without comrpession for supply a line at a Fax and at a modem ? Modem/Fax with VoIP never work ? ___ --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] Asterisk MSOutlook Dialer
Hello everyone, we just wrote a little MSOutlook address book dialer interfaced with Asterisk. It is a small (400k) exe that you need to install. It is completely free to use, either for educational purpose or otherwise. You can download it at http://www.voip.com.sg/voip_products/voip_asterisk_outlook_dialer.html . Please send your comments to me directly as I am the developer for it. With Regards, Sandeep Singhania Lantone Information Systems LLP Tel : SG +65 62271149 (Ext 958) US +1 646 8621550 (ext 958) UK +44 207 0239247 (ext 958) Fax : +65 68750242 Mobile: +65 97471958 Visit our websites to learn more about our products : www.voip.com.sg (Learn more about VOIP and how we can help you implement it) www.mailtracking.com (Track your emails, featured in Channel News Asia and various other publications around the world) www.callaccounting.ws (Home of the world's most popular Call Accounting System proudly developed by us) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: how to define a pilot number
thanks for enlightening. So you mean, if I have 3 lines when the caller dialled the first line and it was busy, the call will be diverted to the next two available lines in random? On 3/27/07, David Cook [EMAIL PROTECTED] wrote: is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Lito I'm assuming you are talking about analog lines as PRI's will do this more-or-less naturally. This is a telco feature as opposed to an Asterisk feature. Here in Bell Canada country they call it Ringer Equivalence. Call your local carrier and they should be able to tell you what they call it in their marketing world. You tell the telco which lines you want the calls to roll to then all three will terminate calls to the pilot number. Now it doesn't work exactly as you had described - it doesn't move the call so as to free up the first port. It merely says the first port is busy and terminates the next call on the next port in sequence. This means you can't count on which line is available at any time. For outbound, you need to put the three lines in an Asterisk group and test the group for availability to select an available line to dial out on. dbc. -- David Cook ___ --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] UK BT PRI
Has anyone got a working zaptel.conf and zapata.conf for a Digium Wildcard TE110P T1/E1 Card. It's connected to a BT ISDN PRI (EuroISDN) with 24 channels. Inbound works fine, but outbound isn't setting CLI (it seems the line supports 6 digit CLI). Inbound CLI works fine. In the dial-plan using Set(CALLERID(num)=123456) then Dial(Zap/g1/01234567||frT) Where 123456 is in the range of BT allocated numbers. Using Asterisk 1.4.1 and Zaptel 1.4.0 Any help appreciated. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using server side phonebook directory with SPA941
On Tue, Mar 27, 2007 at 12:45:44PM +0200, Maxim Veksler wrote:
Hello list,
I got a couple of those wouldn't it be great questions, as following:
1. Is it possible, with asterisk to hold a central phonebook directory
of callers?, so that when this party calls a textual caller ID will
be displayed on the phone display.
Can be done reasonably easily in the dial plan. What I have is quite noddy
but it does the job. In the incoming bits of dial plan where calls come in,
I call this as a macro in the context where incoming calls arrive, before
handing it off to the Dial() bits:
exten = _4535XX,1,Macro(setisdncallerid,${EXTEN},PSTN,9)
What this macro (pasted below) does is allow alpha tagging of incoming
calls, plus some defaulty stuff set by the gateway (caller ID not
present/withheld comes through in my case as either anonymous or just 0 or
00, so this macro tidies this up before passing the call on.)
It also inserts the access digit (9) in front of the caller ID as in my case
outside calls need a 9 prefix. This is just so that call routing works
correctly if people return missed calls/save numbers from the handset etc.
Obviously you will have to tweak this for your setup.
If there is no alpha tag in the DB, it sets some defaulty thing (In my case
PSTN to give some indication where the call is coming from.)
It can also do a CPI tag based on destination number, for queues/group
numbers, so that the alpha tag on the call gets set to something like
Main Number etc. to distinguish a DDI call from a Queue Call.
The database entries look like:
*CLIdatabase put tag 01234567890 Some Name Here
and for CPI (called party) Tag:
*CLIdatabase put 453510 tag Helpdesk
[macro-setisdncallerid]
; ${ARG1} = Called Party Number (XX) as presented from BT.
; ${ARG2} = default tag to add to incoming calls
; ${ARG3} = prefix to insert to incoming CLI
;
; Frobs the incoming caller ID headers how we like it:
exten = s,1,NoOp(macro-setisdncallerid: ${ARG1})
; In my case the internal extension is 7XX where XX is the
; last two digits of the incoming DDI number. This just makes
; it display right in the caller ID:
exten = s,2,Set(DIALED_EXTEN=7${ARG1:-2})
; For cisco phone, set different ring cadence to indicate
; an external call:
exten = s,3,SIPAddHeader(Alert-Info: Bellcore-dr2)
exten = s,4,GotoIf($[ ${CALLERID(num)} = anonymous ]?400)
exten = s,5,GotoIf($[ ${CALLERID(num)} = 0 ]?500)
exten = s,6,GotoIf($[ ${CALLERID(num)} = 00 ]?500)
exten = s,7,GotoIf($[ ${DB(tag/${CALLERID(num)})} != ]?700)
exten = s,8,Set(CALLERID(name)=${ARG2} to ${DIALED_EXTEN})
exten = s,9,Set(CALLERID(num)=${ARG3}${CALLERID(num)})
exten = s,10,Goto(900)
exten = s,400,Set(CALLERID(name)=${ARG2})
exten = s,401,Goto(900)
exten = s,500,Set(CALLERID(num)=unknown)
exten = s,501,Set(CALLERID(name)=${ARG2})
exten = s,502,Goto(900)
exten = s,700,Set(CALLERID(name)=${DB(tag/${CALLERID(num)})})
exten = s,701,Set(CALLERID(num)=${ARG3}${CALLERID(num)})
exten = s,702,Goto(900)
; If there is a CPI tag set, use that: (i.e. SUPPORT)
exten = s,900,GotoIf($[ ${DB(${ARG1}/cpitag)} != ]?950)
exten = s,950,Set(CALLERID(name)=${DB(${ARG1}/cpitag)})
2. How can this be configured with Trixbox, I've looked at the
configuration options - I assume it plays no difference me basing it
on mysql or astdb?
3. What protocol does the phone (Linksys SPA941) talks to the
asterisk server to retrieve this information ?
When an incoming call arrives with asterisk, the SIP headers can be set
appropriately before you present this information to the handset. It's in
the incoming SIP packets to the handset.
4. Has someone done this? What softphone should I use to test it first
(I'm connecting it with outlook, so it has to be win* software)
There are a few to choose from. I use Counterpath's X-Lite client:
http://www.counterpath.com/
Rob
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RE: [asterisk-users] Re: Question about DSP in Digium card
Whether it is IAX, SIP, H323 or ? These are authentication handshakes to establish an rtp stream. SIP = user name and password in a standardized IP packet IAX = same H.323 = same Is also has to do with what codec are supported as well. As far as NAT is concerned! Yep, tell your ISP to forward the authentication port or just junk their gear and get something like a low end Cisco. Or Get IP Phones with STUN (a little pricey) Or Trick Use some type of tunneling gear to an outside IP (outside your NAT) and then bounce your authentication from this new gateway!!! i.e. establish a VPN connection to an outside router from an internal router and drive the call through there. Brad _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of A. Levy Sent: Tuesday, March 27, 2007 6:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Question about DSP in Digium card well, ...,we did not choose SIP because our customers are located behind NAT router (using private IP's) and those routers are not managed by them but by the ISP so it is very difficult to establish full duplex phone calls because you can not initiate voice over ip session from the internet (outside) to LAN side (inside) with private IP's. We could not establish 2-way phone calls, I mean, the conversation is listened in 1-way only. As I mentioned before, we can not configure PAT into the NAT router neither because is handled by the ISP and the passwords are unknown That's why we decided to use IAX instead of SIP, I mean, IAX is more robust than SIP when the NAT router is 3th-party managed and the PAT feature is not enable. On the other and we tested IAX over dialup links and it worked fine Those are the reasons we choose IAX as acess protocol to our SIP/H323 Network. You know, the access networks of the customers are different completely: Private IP Address over DSL lines (NAT Router), Public IP Address over DSL lines, Corporate Networks over dedicated Links (Public and IP Addresses), Dialup links, .. Any comment would be welcomed, thanks a lot Levy.- 2007/3/24, A. Levy [EMAIL PROTECTED]: Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX - ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.- ___ --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] Doorphone
I looked at a call queue, but it didn't seem to work the way I want. Agents need to log into the queue to get calls, seemingly. Of course, I only stopped on the topic for a short period. with the meetme conference, anyone can answer the door from any phone by dialing the conference extension, just not open the door. From: [EMAIL PROTECTED] on behalf of Ola Lidholm Sent: Mon 3/26/2007 7:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Doorphone On 26 mar 2007, at 22.17, Ray Wadkins wrote: We have a doorphone device that's connected to our PBX. Currently, there's a special meetme conference that the phone connects to when the visitor presses zero. Users in the office can dial the meetme conference and get connected. The problem is that we can't send DTMF signals to the door to open it, because the meetme app seems to capture them. I had the bright idea to set up a virtual extension that would just ring, virtually. Then we could use call pickup to snag the call at an extension and be able to open the door. Unfortunately, I can't figure out how to get that to work. Wait(30) and Answer (3) don't seem to allow call pickup to snag the extension. Any suggestions? Hi Ray, I can't really understand why you want to use a meetme conference? Why not use a call queue instead? /Ola Lidholm [EMAIL PROTECTED] Whatever one man is capable of conceiving, other men are able to achieve. - Jules Verne. ___ --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 winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: how to define a pilot number
Lito Lampitoc wrote: thanks for enlightening. So you mean, if I have 3 lines when the caller dialled the first line and it was busy, the call will be diverted to the next two available lines in random? I don't think it's random. I think its just sequential. If main line is busy, try second. If that is unavailable, then try third in sequence, etc. It's called rollover here. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-registration ?
2. Is possible to do the same with SIP hardphones ? Some hardphones support registering to multiple sip accounts from one phone. (as indeed do some softphones) Is that what you want ? Yes but my question is : Is it possible to register 2 accounts for the same user and hardphone within the same Asterisk server ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Question about DSP in Digium card
You've got a decent server. Generally the limiting factor for the number of simultaneous calls is more about server memory. That server could likely handle 124 simultaneous calls, but you would be prudent to double that memory size. Make sure you are running at 100 full especially if you are using G.711. 10 Full uplinks won't cut it if you are running that kind of bandwidth. As for the DSP, you are right to be concerned about the Digium cards, but not because of the DSP. The DSP is not where you will run into problems. Digium cards feature 2 year old circuitry and do not play well with other devices. You have to take care not to share interrupts with any components that may be active on that system. Sharing an IRQ between a Digum card and an Ethernet card would certainly spell disaster in my experience. From personal experience, I no longer use Digium hardware since I could rarely push a quad port card to more than 13 channels per T1 circuit without the card failing miserably. HDLC aborts abound. For now, I only use Sangoma cards. These don't have the IRQ issues and I have had no problems pushing their cards to their maximum. I recommend echo canceller enabled cards for any T1/E1's you may use that are not long distance carrier lines. Good luck, hope this helps with your capacity planning. - SG ## 2007/3/24, A. Levy [EMAIL PROTECTED]: Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX - ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.- ___ --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] Multi-registration ?
Asterisk can handle multiple registrations for the same account. Both should ring when calls come in. If you are using the same account for both line appearances, theoretically it should work on a phone like a Cisco 7960, but it would behave strangely when calls came in. Both line appearances would indicate an inbound call. If you are using two different accounts, there will be no problems at all. Each line appearance would register and could receive calls on either. Good luck, SG -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http:// http://VoIPSecurityTraining.com VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Tuesday, March 27, 2007 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-registration ? 2. Is possible to do the same with SIP hardphones ? Some hardphones support registering to multiple sip accounts from one phone. (as indeed do some softphones) Is that what you want ? Yes but my question is : Is it possible to register 2 accounts for the same user and hardphone within the same Asterisk server ? ___ --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: Question about DSP in Digium card
In article [EMAIL PROTECTED], Salvatore Giudice [EMAIL PROTECTED] wrote: From personal experience, I no longer use Digium hardware since I could rarely push a quad port card to more than 13 channels per T1 circuit without the card failing miserably. HDLC aborts abound. This usually happens if you have left the software echo canceller enabled, because all the zaptel echo cancellation happens in the interrupt service routine (!!!). With echo cancellation disabled, I have found that I can fill a TE405P/TE410P with calls quite happily on a single P4 at 2.8GHz, and no HDLC aborts. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cisco 7905
How to configure cisco 7905 with asterisk ,if you please can send me step by step configuration steps . Thanks Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --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: asterisk-users Digest, Vol 32, Issue 106
Lito Lampitoc wrote: thanks for enlightening. So you mean, if I have 3 lines when the caller dialled the first line and it was busy, the call will be diverted to the next two available lines in random? I don't think it's random. I think its just sequential. If main line is busy, try second. If that is unavailable, then try third in sequence, etc. It's called rollover here. Correct, its in the sequence you told the carrier you want. Caveat, You _can_ have contention with analog lines. Meaning someone calling in at precisely the same time as someone calling out - not often, but it will happen. To help aleviate this, get the carrier to roll the lines 1-2-3 and outbound you pick the lines 3-2-1. - dbc ___ --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] IAX Experiences [WAS: Question about DSP in Digium card]
Hi Steve - Sorry for the dupe, but since this is now way off-thread, I thought I'd create a new one (and correct my spelling mistake). Just my personal experience, but I do not find IAX to be very reliable. Is there any particular reason you are not using SIP? I'm curious as to your negative experiences with IAX. I generally use it for multi-office installations, and have had good experiences. What reliability issues did you see? Jitter? Drops? Thanks, Noah ___ --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] AOC billing
Hello, is there someone who knows if I can use AOC for billing in Asterisk? I mean: let's say I have an external SIP device that produces AOC data. This device connects me to the telco network. Can Asterisk, if connected via SIP with this device, collect AOC data and put it in the CDR records? If not, which is the right way to use AOC for billing? Thanks a lot Stefano Corsi -- Stefano Corsi www.floo.it via della Fiera, 1 57029, Venturina - Campiglia Marittima (LI) Tel. 0565-836130 - Fax. 0565-836143 Cell. 320-3484294 ___ --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] cisco 7905
Khaled, Check this URL http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a 0080094584.shtml _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Tuesday, March 27, 2007 4:54 PM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject: [asterisk-users] cisco 7905 How to configure cisco 7905 with asterisk ,if you please can send me step by step configuration steps . Thanks Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: how to define a pilot number
Lee Jenkins wrote: Lito Lampitoc wrote: thanks for enlightening. So you mean, if I have 3 lines when the caller dialled the first line and it was busy, the call will be diverted to the next two available lines in random? I don't think it's random. I think its just sequential. If main line is busy, try second. If that is unavailable, then try third in sequence, etc. It's called rollover here. It is also called a hunt group. The telco will usually roll the call the the next available line in the huntgroup. However they can also roll to the longest idle line. This can help in modem pools where if a modem connected to the 2nd (or whatever) line of the hunt group dies then the line won't be busy and the call won't roll over. longest idle can at least let most callers get thru in the event of a failure on one of the modems. When dealing with analog lines you have to be concerned about glare. Glare happens when the PBX picks up a line at the exact moement the telco sends a call to the line. One way to do this is to have the PBX hunt from the top of the hunt group and work its way down, and have the telco hunt from the bottom up. ___ --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] Doorphone
On 3/27/07, Ray Wadkins [EMAIL PROTECTED] wrote: I looked at a call queue, but it didn't seem to work the way I want. Agents need to log into the queue to get calls, seemingly. Of course, I only stopped on the topic for a short period. with the meetme conference, anyone can answer the door from any phone by dialing the conference extension, just not open the door. You can have static agents so they don't have to login, check http://www.voip-info.org/wiki-Asterisk+call+queues Wondering why you don't just dial multiple-phones, like this Dial(SIP/7001SIP/7002SIP/7003) The first one that answer the call is the lucky one. That way, your DTMF signals would work. hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue App - Free agent and waiting calls
Any news of this behavior? bweschke, could you work on this bug?? On 3/19/07, equis software [EMAIL PROTECTED] wrote: Please send me any news about this or the bug number. Thanks for your time. On 3/19/07, BJ Weschke [EMAIL PROTECTED] wrote: On 3/19/07, equis software [EMAIL PROTECTED] wrote: Asterisk 1.4 I have strategy= leastrecent and autofill = yes I have 2 agents, one is answering a call and the other is free and have some calls waiting in the queue. Only when the first agent hangup the second agent receive the first call in the queue. It happends some times. I believe there is a bug open in Mantis on this, and I intend to reproduce and start working on it this week to get a resolution. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Doorphone
Responsibility for answering the door is shared by the entire office. But A) noone wants their phone to ring, there's a door chime) and B) noone specific will accept responsibility for answering the door. So, we need a solution that follow I'm answering the door now, these are the buttons I push. From: [EMAIL PROTECTED] on behalf of Time Bandit Sent: Tue 3/27/2007 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Doorphone On 3/27/07, Ray Wadkins [EMAIL PROTECTED] wrote: I looked at a call queue, but it didn't seem to work the way I want. Agents need to log into the queue to get calls, seemingly. Of course, I only stopped on the topic for a short period. with the meetme conference, anyone can answer the door from any phone by dialing the conference extension, just not open the door. You can have static agents so they don't have to login, check http://www.voip-info.org/wiki-Asterisk+call+queues Wondering why you don't just dial multiple-phones, like this Dial(SIP/7001SIP/7002SIP/7003) The first one that answer the call is the lucky one. That way, your DTMF signals would work. hth ___ --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 winmail.dat___ --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] TDM400p reliability
What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? joe a. ___ --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] just call to user
hello i have installed Asterisk on a Debian machine by apt-get install asterisk I only want to call a user inside the LAN, what files I have to edit??? sip.conf??? thanks for all___ --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] TDM400p reliability
Joe Acquisto wrote: What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? joe a. I'm using one at home in that exact configuration. I have a POTS line and a dock-n-talk cell station on the FXOs and two cordless on the FXSs. I don't recall ever experiencing any failures with it. But, being it's a home PBX it doesn't get a lot of traffic either. -Dave ___ --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] just call to user
Asterisk isn't a simple apt-get and run type program...have a look at the asterisk wiki for help getting started. There's a lot to configure MD _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josu Lazkano Lete Sent: Tuesday, March 27, 2007 11:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] just call to user hello i have installed Asterisk on a Debian machine by apt-get install asterisk I only want to call a user inside the LAN, what files I have to edit??? sip.conf??? thanks for all ___ --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: SIP/IAX peers UNREACHABLE and audio loss
Hi all, I made some tests under heavy network load generated artificially moving files form server to server I noticed a 3% packet loss in ping -f response form server involved in big data transfer (1 GB files through http) I changed the network switch with a Cisco Catalyst 2950 and the packet loss with pings disapperead but the problem with REACHABLE / UNREACHABLE peers remains... I did one more simple test While Asterisk is stating the peer is UNREACHABLE I can ping (even -f) it without problem and without packet loss. Could it be a problem in Asterisk ? I'm using 1.2.13 on a gentoo Kernel 2.6.20 Tnx again for help Edoardo Edoardo Serra ha scritto: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Multi-registration ?
On 27/03/07, Salvatore Giudice [EMAIL PROTECTED] wrote: Asterisk can handle multiple registrations for the same account. Both should ring when calls come in. No it can't - the latest registration 'wins'. To achieve simutaneous ringing of more than one phone (hard or soft), you need a SIP account for each and an entry in the dialplan which rings both. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] Asterisk 1.4 and chan_misdn
Hi Pierre and the list, I have the habit to do like this after having compiled Zaptel and Libpri : cd /usr/src/ wget http://www.misdn.org/downloads/mISDN.tar.gz wget http://www.misdn.org/downloads/mISDNuser.tar.gz tar xzf mISDN.tar.gz tar xzf mISDNuser.tar.gz cd mISDN-1_1_1 make install cd ../mISDNuser-1_1_1 make install Move the modules which are in the bad directory : mkdir /lib/modules/`uname -r`/extra cp /lib/modules/extra/*.* /lib/modules/`uname -r`/extra Make the mISDN config files : /etc/init.d/misdn-init config Start mISDN : /etc/init.d/misdn-init start Go ahead and compile Asterisk : cd /usr/src/asterisk-1.4 ./configure make menuselect ; choose your options ! make;make install I hope this help ! Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Pierre Burton Envoyé : mardi 27 mars 2007 10:09 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Asterisk 1.4 and chan_misdn Hi, you also need mISDNuser. After that make clean make install you'll have access to chan_misdn. Regards. Pierre Administrator TOOTAI wrote: Hi list, I installed a fresh Debian/Etch with Asterisk 1.4 and Zaptel 1.4 from SVN for 2 Digium B410P card. I ran configure in Asterisk dir, went in zaptel dir and: make, make install, make b410p. Everything is ok. Now I want to compile Asterisk but can't activate the chan_misdn channel which depends on -from menuselect- isdnnet(E), misdn(E), suppserv(E) When I made the make b410p, all the misdn stuff was downloaded from digium's ftp. Also, running /etc/init.d/misdn-init --scan show me the 2 cards I have, /etc/init.d/misdn-init --config prepare me the misdn.conf and after a /etc/init.d/misdn-init start I see: mISDN_dsp 191656 0 mISDN_capi 88716 0 mISDN_l2 34452 0 mISDN_l1 11036 0 mISDN_core 71360 6 mISDN_dsp,hfcmulti,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1 kernelcapi 44576 2 mISDN_capi,capi My questions: why Asterisk doesn't want to let me activate the misdn channel? Is misdn ready for 1.4? Thanks for any hint ___ --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] P-Asserted-Identify or Remote-Party-ID, or both?
For INBOUND calls, does Asterisk support P-Asserted-Identify or Remote-Party-ID, or does it support both? Again, this is for INBOUND only. I know how to add those headers for outbound calls. My guess from what I have seen is that it supports both, but I wanted to check with the list. ___ --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] TDM400p reliability
On Tue, Mar 27, 2007 at 11:15:57AM -0400, Joe Acquisto wrote: What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. I have a 3 FXS and 1 FXO. Apart from the FXO blowing the first time the line rang, which my supplier quickly replaced, I have had no real problems, other than the odd power alarm on the FXS modules. -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --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] TDM400p reliability
At 08:25 AM 3/27/2007, you wrote: What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? I have an TDM04 that has been working perfectly for over a year. I only have it connected to phone lines, all my phones are SIP and the only problem ever was echo and the HPEC has fixed that. Ira ___ --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] TDM400p reliability
What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? We have quite a few dotted around clients' places to handle emergency calls (and a few other call types we can't run through IAX/SIP to our carrier). I've never had one actually *fail* in the last few years, though they can be something of a nightmare to get right with echo issues and the like. Most of ours are in either 1 or 2 FXO configuration - only a couple have FXS modules on them. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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] TDM400p reliability
We have had much better success with the Sangoma A200 Excellent support from Sangoma Works in all modern motherboards - no try another motherboard answers from support Expandable, if needed, to 24 ports Lower price per port, depending on your supplier. John Novack Joe Acquisto wrote: What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 106
I think it is called hunt group in my neck of the woods. -Original Message- From: David Cook [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 27, 2007 6:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 106 Lito Lampitoc wrote: thanks for enlightening. So you mean, if I have 3 lines when the caller dialled the first line and it was busy, the call will be diverted to the next two available lines in random? I don't think it's random. I think its just sequential. If main line is busy, try second. If that is unavailable, then try third in sequence, etc. It's called rollover here. Correct, its in the sequence you told the carrier you want. Caveat, You _can_ have contention with analog lines. Meaning someone calling in at precisely the same time as someone calling out - not often, but it will happen. To help aleviate this, get the carrier to roll the lines 1-2-3 and outbound you pick the lines 3-2-1. - dbc ___ --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] TDM400p reliability
There are downsides to the A200 (which I have had very good luck with as well and highly recommend, don't get me wrong). You have to install FXO or FXS ports in pairs, you can't do 3 FXO and 1 FXS for example. The other is having to manage one more set of drivers (wanpipe). Not a big deal though and I've never had any problems doing so. To start a bunny trail, your try another motherboard comment made me wonder about this new VoiceBus technology mentioned on the new TDM800 and TE120P cards. What exactly is it? Is is just a new PCI interface on the card? What makes it work so much better than the other cards? -Dave John Novack wrote: We have had much better success with the Sangoma A200 Excellent support from Sangoma Works in all modern motherboards - no try another motherboard answers from support Expandable, if needed, to 24 ports Lower price per port, depending on your supplier. John Novack Joe Acquisto wrote: What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] P-Asserted-Identify or Remote-Party-ID, or both?
On 3/27/07, Matt [EMAIL PROTECTED] wrote: For INBOUND calls, does Asterisk support P-Asserted-Identify or Remote-Party-ID, or does it support both? Again, this is for INBOUND only. I know how to add those headers for outbound calls. My guess from what I have seen is that it supports both, but I wanted to check with the list. Matt, I don't know if it supports PAI yet, but to get RPID you need to set: trustrpid=yes in [general] or for the peer. -- Kristian Kielhofner ___ --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] Park No Announce?
We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the announce part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken ___ --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] Park No Announce?
Ken, Just curious, how did you make the Granstream's lights light up when someone is parked? On 3/27/07, Ken Williams [EMAIL PROTECTED] wrote: We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the announce part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken ___ --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] Cisco 30VIP Phone
Is anyone else on the list using Cisco 30VIP phones with the chan_skinny driver? I have tried to catch the one of the developers on the chat relay, but cannot seem to get anywhere. I am trying to understand how the soft buttons are setup. They are apparently hard-coded into the chan_skinny.c module. Specifically, I am looking for how the code relates to the actual layout of the buttons on the phone. So far, I cannot even get the buttons that are in the code by default to work properly. I have several of these phones up and registered with *. The dialpads work fine. But other buttons do not. Thanks Chris On 3/23/07, Chris Nighswonger [EMAIL PROTECTED] wrote: On 3/23/07, Chris Nighswonger [EMAIL PROTECTED] wrote: I have three registering with * and having basic functionality. I am at a loss to know how to program the buttons (other than dtmf, hold, mute, spkr). Here is what the * console shows when one of the phones registers: -- Starting Skinny session from 192.168.0.70 -- Device 'SEP000196C00CDC' successfully registered Device capability set to '12' Adding button: 9, 1 Adding button: 1, 0 Adding button: 15, 0 Adding button: 126, 0 Adding button: 5, 0 Adding button: 125, 0 It appears that * is setting up some buttons. But where it is getting the config info, I don't know. Sorry for answering my own post, however it may help someone else: Soft button configuration is set in skinny.c I'm still looking for some explaination of the logic and sytax of setting them. Chris -- Chris Nighswonger Network Systems Director Foundations Bible College Seminary www.foundations.edu www.fbcradio.org [EMAIL PROTECTED] V:910-892-8761 C:919-820-5473 - NOTICE: The information contained in this electronic mail message is intended only for the use of the intended recipient, and may also be protected by the Electronic Communications Privacy Act, 18 USC Sections 2510-2521. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please reply to the sender, and delete the original message. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Park No Announce?
In the basic settings, I setup the Multi-Purpose Key to use Asterisk BLF and assigned it the parking lot extension (201 in our case, 701 by default iirc). I then added hints in the extensions.conf for the parking lot extensions: exten = 201,hint,park:[EMAIL PROTECTED] exten = 201,1,Wait(1) exten = 201,2,ParkedCall(201) Of course in features.conf I've defined the context as parkedcalls (context = parkedcalls). Hope this helps, it took a lot of piecing together other examples to get it to come together so let me know if you need more help. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, March 27, 2007 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Park No Announce? Ken, Just curious, how did you make the Granstream's lights light up when someone is parked? On 3/27/07, Ken Williams [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the announce part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken ___ --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] Park No Announce?
So are you running BRIStuff for this to work? On 3/27/07, Ken Williams [EMAIL PROTECTED] wrote: In the basic settings, I setup the Multi-Purpose Key to use Asterisk BLF and assigned it the parking lot extension (201 in our case, 701 by default iirc). I then added hints in the extensions.conf for the parking lot extensions: exten = 201,hint,park:[EMAIL PROTECTED] exten = 201,1,Wait(1) exten = 201,2,ParkedCall(201) Of course in features.conf I've defined the context as parkedcalls (context = parkedcalls). Hope this helps, it took a lot of piecing together other examples to get it to come together so let me know if you need more help. -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Matt *Sent:* Tuesday, March 27, 2007 1:59 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Park No Announce? Ken, Just curious, how did you make the Granstream's lights light up when someone is parked? On 3/27/07, Ken Williams [EMAIL PROTECTED] wrote: We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the announce part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Park No Announce?
Ken Williams wrote: We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the announce part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken If I send the announce to an invalid extension it still seems to park the call fast enough. I suppose I could create an extension that just answers and hangs up to get rid of the warning messages. ___ --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] Park No Announce?
Paul wrote: Ken Williams wrote: We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the announce part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken If I send the announce to an invalid extension it still seems to park the call fast enough. I suppose I could create an extension that just answers and hangs up to get rid of the warning messages. Okay, I am now specifying Local/parkannounce as the announce extension. That extension answers and hangs up. The calls park and MOH starts immediately. ___ --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] Park No Announce?
No, this is a pretty plain vanilla setup, never touched BRIStuff. My features.conf (which defines the parkandannounce app) looks like: [general] parkext = 200 ; What extension to dial to park parkpos = 201-210 ; What extensions to park calls on. These needs$ ; numeric, as Asterisk starts from the start po$ ; and increments with one for the next parked c$ context = parkedcalls ; Which context parked calls are in parkingtime = 75 ; Number of seconds a call can be parked for From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, March 27, 2007 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Park No Announce? So are you running BRIStuff for this to work? On 3/27/07, Ken Williams [EMAIL PROTECTED] wrote: In the basic settings, I setup the Multi-Purpose Key to use Asterisk BLF and assigned it the parking lot extension (201 in our case, 701 by default iirc). I then added hints in the extensions.conf for the parking lot extensions: exten = 201,hint,park:[EMAIL PROTECTED] exten = 201,1,Wait(1) exten = 201,2,ParkedCall(201) Of course in features.conf I've defined the context as parkedcalls (context = parkedcalls). Hope this helps, it took a lot of piecing together other examples to get it to come together so let me know if you need more help. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, March 27, 2007 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Park No Announce? Ken, Just curious, how did you make the Granstream's lights light up when someone is parked? On 3/27/07, Ken Williams [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the announce part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Park No Announce?
The problem isn't on the outside phone, it's on the inside. An outside caller already gets MOH immediately, the problem comes in waiting for the TRANSFER to complete. What we're doing to park a call is hitting TRNF on the GXP-2000 followed by 200 (the park extension). The phone then says 'TRANSFERRING' and waits 2-3 seconds before saying TRANSFER SUCCESSFUL. The time that it's waiting is exactly how long it takes the system to say TWO ZERO ONE. I'd like to disable the Announce part so it just transfers. Hope that's a little clearer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Tuesday, March 27, 2007 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Park No Announce? Paul wrote: Ken Williams wrote: We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the announce part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken If I send the announce to an invalid extension it still seems to park the call fast enough. I suppose I could create an extension that just answers and hangs up to get rid of the warning messages. Okay, I am now specifying Local/parkannounce as the announce extension. That extension answers and hangs up. The calls park and MOH starts immediately. ___ --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] Erased log files
People, I've erased the *messages* and *full *files in /var/log/asterisk/. I've already created other files and changed the owner, etc, and permissions: *-rw-r--r-- 1 asterisk asterisk 0 Mar 2 16:01 event_log -rw-r--r-- 1 asterisk asterisk 1514385 Mar 27 18:15 full -rw-r--r-- 1 asterisk asterisk 396170 Mar 27 18:20 messages -rw-r--r-- 1 asterisk asterisk1102 Mar 3 18:08 queue_log * But, the asterisk is not writing in these files anymore. Is there a solution? Thanks. -- Abraços Luis Claudio ___ --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] Erased log files
logger reload at the CLI Luis Claudio Santos wrote: People, I've erased the messages and full files in /var/log/asterisk/. I've already created other files and changed the owner, etc, and permissions: -rw-r--r-- 1 asterisk asterisk 0 Mar 2 16:01 event_log -rw-r--r-- 1 asterisk asterisk 1514385 Mar 27 18:15 full -rw-r--r-- 1 asterisk asterisk 396170 Mar 27 18:20 messages -rw-r--r-- 1 asterisk asterisk1102 Mar 3 18:08 queue_log But, the asterisk is not writing in these files anymore. Is there a solution? Thanks. -- Abraços Luis Claudio ___ --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] TDM400p reliability
Dave Fullerton wrote: To start a bunny trail, your try another motherboard comment made me wonder about this new VoiceBus technology mentioned on the new TDM800 and TE120P cards. What exactly is it? Is is just a new PCI interface on the card? What makes it work so much better than the other cards? It is exactly that; a different PCI interface that is far, far less likely to experience PCI compatibility issues than the TigerJet interface on the TDM400P and TE110P. In addition to that, recent driver improvements in Zaptel have also solved many interrupt-related problems even with the TDM400P and TE110P. ___ --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] Doorphone
On 27 mar 2007, at 15.15, Ray Wadkins wrote: I looked at a call queue, but it didn't seem to work the way I want. Agents need to log into the queue to get calls, seemingly. Of course, I only stopped on the topic for a short period. with the meetme conference, anyone can answer the door from any phone by dialing the conference extension, just not open the door. In queue.conf (or is it called queues.conf?) you can set up a call queue with all your phones already in it. Which will mean that if you pass the incoming call to that queue all phones will be ringing until one person picks it up. At my work we have it set up like that. And additionally, people can join or leave the call queue by dialing certain extensions on their phones, which can be convenient when people do not want to be disturbed. I do not understand exactly how you mean your system works, how does the users know when someone is at the door? Since no phone is ringing it seems to me like a guessing game to know when they need to dial in to the meetme to open the door? Do you have free sight to the entrance door so that you can see if someone is already there? /Ola From: [EMAIL PROTECTED] on behalf of Ola Lidholm Sent: Mon 3/26/2007 7:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Doorphone On 26 mar 2007, at 22.17, Ray Wadkins wrote: We have a doorphone device that's connected to our PBX. Currently, there's a special meetme conference that the phone connects to when the visitor presses zero. Users in the office can dial the meetme conference and get connected. The problem is that we can't send DTMF signals to the door to open it, because the meetme app seems to capture them. I had the bright idea to set up a virtual extension that would just ring, virtually. Then we could use call pickup to snag the call at an extension and be able to open the door. Unfortunately, I can't figure out how to get that to work. Wait(30) and Answer (3) don't seem to allow call pickup to snag the extension. Any suggestions? Hi Ray, I can't really understand why you want to use a meetme conference? Why not use a call queue instead? /Ola Lidholm [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] ztdummy and MOH
It would make more sense if you posted the musiconhold.conf file and stated if you did or didn't install the asterisk_addons package with mp3 support. On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct. Please help. 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 ___ --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] ARI with * 1.4.2 won't display recordings
Evnin' Now I tracked my problem down why ARI won't display most of the recordings... It write a recording for examples as: 1175031785-SIP-0615000995-0872a000.wav But it writes to the field uniqieid into MySQL database as: 1175031779.16 WHen I overwrite the uniqueid field with the value from the recording file, the recording is playable within ARI: 1175031785 Any idea why the uniqieid and the ID used for creating the recording files are always different? cheers rick ___ --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] vzaphfc installation...
On Tue, Mar 27, 2007 at 11:20:49AM +0200, Mauro Zanin wrote: Hi everybody, does anybody knows how to install and configure VZAPHFC? Basically something of the sort of: Copy the sources of vzaphfc as a subdirectory vzaphfc under the sources of zaptel . Add to Makefile.linux26 the line: obj-m += vzaphfc/ (or get the zaptel/zaptel-source packages from Debian) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] TDM400p reliability
Actually I think the additional wanpipe drivers are a major plus. For PRI troubleshooting I even think there are tools that are not present in Zaptel, or are much harder to setup (vs wanpipemon -g). Commitment to multiple protocols, applications and platforms is also another plus. Sangoma supports many configurations I doubt Digium ever will. On 3/27/07, Dave Fullerton [EMAIL PROTECTED] wrote: There are downsides to the A200 (which I have had very good luck with as well and highly recommend, don't get me wrong). You have to install FXO or FXS ports in pairs, you can't do 3 FXO and 1 FXS for example. The other is having to manage one more set of drivers (wanpipe). Not a big deal though and I've never had any problems doing so. To start a bunny trail, your try another motherboard comment made me wonder about this new VoiceBus technology mentioned on the new TDM800 and TE120P cards. What exactly is it? Is is just a new PCI interface on the card? What makes it work so much better than the other cards? -Dave John Novack wrote: We have had much better success with the Sangoma A200 Excellent support from Sangoma Works in all modern motherboards - no try another motherboard answers from support Expandable, if needed, to 24 ports Lower price per port, depending on your supplier. John Novack Joe Acquisto wrote: What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inbound Voice Quality - Speed Change
Many times the speed of an inbound voice call changes. It's similiar to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible. A speed change is the best way to describe it, seems like the voice packets are being played out too fast. Can anyone explain what might cause this? It doesn't always happen, and seem unpredictable. Thanks, Jim ___ --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] Doorphone vs. Grandstream BT101
This is the simplest solution I can think of: http://www.smarthome.com/5070cw.html On 3/26/07, Jay Milk [EMAIL PROTECTED] wrote: Steve Totaro wrote: Just get a Grandstream ATA and a handset with no buttons. So simple. That doesn't really meet my needs -- I want to be able to dial-out, and have the person on the other end simply be able to push a button to ring the doorbell. The doorbell button requirement stems from the eternal hope that someday DHL drivers will be trained to push it just before or after they slam-dunk that box marked fragile, so I can get this box of broken computer parts out of the pouring rain when it arrives, and won't have to file the insurance claim days later when I find the rain-soaked cardboard blob near the culvert. Sorry, I got sidetracked there. What I mean is, I'd like to keep the doorbell button so that Fedex and UPS drivers can continue to ring it and leave when they deliver something -- I think having to pick up a crusty, dirty receiver might be a deterrent even to those 99.9% of folks who are better trained than a DHL driver. Shoot, went there again. No I'm not bitter. ___ --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] Couldn't load variables.txt?aldope=xxxxx
HI!!!Sorry this post about FOP but it's important. Ive installed asterisk and freepbx. the interface of FreePBX works fine, but when i acesse FOP (Flash Operator Panel) i get this error: Couldn't load variables.txt?aldope=x I search in the google and see a sugestion to edit line flash_dir=/var/www/html/panel/flash in file op_server.cfg. Any Sugestion please? -- Carlos Jerónimo ___ --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] Multi-registration ?
On 3/26/07, Olivier [EMAIL PROTECTED] wrote: Hello, 1. Is it possible to install several SIP softphones on the same PC, Yes. You can even install for example two softphones for Windows, two for Linux and two for MacOSX (two is an imaginary number you can have six on Windows, 27 on Linux and none on Mac for example. have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? yes, but why not use a phone with multiple lines Its liks saying you can buy a Cisco hardphone with 10 lines but then only use 1 line on each and buy 10 phones -- for the same person. 2. Is possible to do the same with SIP hardphones ? A sip hardphone is nothing but specialized computer running special OS and softphone. Besides the fact that one is hardware and the other is software from a technical perspective is the same. So I suppose your question is if there are hardphones with that sort of software available and the answer is yes. But what about several registrations from the same hardphone to the same Asterisk server ? It might be a simple way to allocate specific behaviour : For instance, someone could be both seen as call center staff and post-sales department member. From dialplan design and maintenance perspective, do you think it would be more simple to associate registration and business assignment ? Think about this as an analogy. If the agent become the user ID then why are you going to call the agent John also Bob and James? That makes no sense. If a user is getting a call you have to route that call somehow. You know the user wants post-sales support and you have a list of agents that can handle post sales support. Why are you going to reconfigure everything? So what if that user moves to a different desk you need to configure that new phone to have an extra line? If you for example know John uses agent login why not send the call to John instead of his line. Am I making sense here? ___ --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] Multi-registration ?
Sorry. My mistake. I was thinking of SER. You are quite right. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Bowyer Sent: Tuesday, March 27, 2007 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-registration ? On 27/03/07, Salvatore Giudice [EMAIL PROTECTED] wrote: Asterisk can handle multiple registrations for the same account. Both should ring when calls come in. No it can't - the latest registration 'wins'. To achieve simutaneous ringing of more than one phone (hard or soft), you need a SIP account for each and an entry in the dialplan which rings both. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Inbound Voice Quality - Speed Change
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: Many times the speed of an inbound voice call changes. It's similiar to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible. A speed change is the best way to describe it, seems like the voice packets are being played out too fast. Can anyone explain what might cause this? It doesn't always happen, and seem unpredictable. This sounds like a timing issue. Do you have zaptel cards installed or ztdummy? If zaptel cards, are you getting your timing from the telco or providing your own? I shut off the line that was providing timing from the telco on another system (not asterisk) and all heck broke loose. I had called to cancel the line a few days earlier and it took a couple days for them to cancel it, so basically had forgotten about it. Took me awhile to figure that one out. ___ --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] Macro Dial - External DID
I am using the sample (slightly modified) macro for dialing phones. My
extensions are in the 2000 range. Each extension has it's own
external DID. Is there any way to have the macro look up the DID to
be used for the extension and set the DID as the callerid? Maybe from
a flat file somewhere? Or is there a better way to do this???
I know I can use callerid in sip.conf, but I only want the DID used
when the call goes external.
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED]) ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED]) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user
into VoicemailMain
[phones]
exten = _2XXX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
DID example:
2001 = 5552871701
2002 = 5552871702
2003 = 5552871703
Thanks!
--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
___
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[asterisk-users] Re: Inbound Voice Quality - Speed Change
Lacy, I don't have any zaptel cards installed. I do however have ztdummy installed. Is there some tweaks to ztdummy which I might need? Is there a special kernel setting which ztdummy requires? Jim Lacy Moore - Aspendora wrote: On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: Many times the speed of an inbound voice call changes. It's similiar to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible. A speed change is the best way to describe it, seems like the voice packets are being played out too fast. Can anyone explain what might cause this? It doesn't always happen, and seem unpredictable. This sounds like a timing issue. Do you have zaptel cards installed or ztdummy? If zaptel cards, are you getting your timing from the telco or providing your own? I shut off the line that was providing timing from the telco on another system (not asterisk) and all heck broke loose. I had called to cancel the line a few days earlier and it took a couple days for them to cancel it, so basically had forgotten about it. Took me awhile to figure that one out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Inbound Voice Quality - Speed Change
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: I don't have any zaptel cards installed. I do however have ztdummy installed. Hmm... Not sure. But this really sounds like ztdummy is not working correctly. Hopefully someone else can jump in here. The only system I've ever done without a zaptel card have been lab systems, and they have worked as far as I can tell, with ztdummy. WHat version of Asterisk and what version of zaptel are you running? ___ --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: Inbound Voice Quality - Speed Change
Lacy, I'm using asterisk 1.4.2 and zaptel 1.4.1. I read the READMEs again. I believe I need to change my kernel RTC to 1000HZ. Also, I didn't have enhanced_real_time clock enabled, as such, ztdummy wasn't loading properly. I have rebuilt and started testing again. Thanks for the replies!! Jim Lacy Moore - Aspendora wrote: On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: I don't have any zaptel cards installed. I do however have ztdummy installed. Hmm... Not sure. But this really sounds like ztdummy is not working correctly. Hopefully someone else can jump in here. The only system I've ever done without a zaptel card have been lab systems, and they have worked as far as I can tell, with ztdummy. WHat version of Asterisk and what version of zaptel are you running? ___ --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] Park No Announce?
I couldn't find a switch, so I commented line 426 out of res_features.c and recompiled - instant transfer now on Grandstream phones. Below is the line for future reference. ast_say_digits(peer, pu-parkingnum, , peer-language); From: [EMAIL PROTECTED] on behalf of Ken Williams Sent: Tue 3/27/2007 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Park No Announce? The problem isn't on the outside phone, it's on the inside. An outside caller already gets MOH immediately, the problem comes in waiting for the TRANSFER to complete. What we're doing to park a call is hitting TRNF on the GXP-2000 followed by 200 (the park extension). The phone then says 'TRANSFERRING' and waits 2-3 seconds before saying TRANSFER SUCCESSFUL. The time that it's waiting is exactly how long it takes the system to say TWO ZERO ONE. I'd like to disable the Announce part so it just transfers. Hope that's a little clearer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Tuesday, March 27, 2007 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Park No Announce? Paul wrote: Ken Williams wrote: We're using Grandstream GXP-2000 with programmed buttons to the first 5 parking lot extensions. When a call is parked, whichever parking lot extension it's parked on lights up red. We've never used the announce part and I'm wondering if there's an option I can't seem to find to disable the announce so the transfer happens faster. Thanks for any help, Ken If I send the announce to an invalid extension it still seems to park the call fast enough. I suppose I could create an extension that just answers and hangs up to get rid of the warning messages. Okay, I am now specifying Local/parkannounce as the announce extension. That extension answers and hangs up. The calls park and MOH starts immediately. ___ --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 winmail.dat___ --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: Inbound Voice Quality - Speed Change
Lacy, How I have this: ztdummy 4424 0 rtc11156 1 ztdummy zaptel178084 1 ztdummy crc_ccitt 2016 1 zaptel ztdummy is loaded, and * is running, however, I would have expected ztdummy to be used by at least something. Does this look normal? Jim Jim Duda wrote: Lacy, I'm using asterisk 1.4.2 and zaptel 1.4.1. I read the READMEs again. I believe I need to change my kernel RTC to 1000HZ. Also, I didn't have enhanced_real_time clock enabled, as such, ztdummy wasn't loading properly. I have rebuilt and started testing again. Thanks for the replies!! Jim Lacy Moore - Aspendora wrote: On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: I don't have any zaptel cards installed. I do however have ztdummy installed. Hmm... Not sure. But this really sounds like ztdummy is not working correctly. Hopefully someone else can jump in here. The only system I've ever done without a zaptel card have been lab systems, and they have worked as far as I can tell, with ztdummy. WHat version of Asterisk and what version of zaptel are you running? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Doorphone
Ray Wadkins wrote: I had the bright idea to set up a virtual extension that would just ring, virtually. Then we could use call pickup to snag the call at an extension and be able to open the door. Unfortunately, I can't figure out how to get that to work. Wait(30) and Answer(3) don't seem to allow call pickup to snag the extension. Any suggestions? Sure, try something like this: [doorcom] exten = s,1,Dial(Local/ringforever) [ringforever] exten = s,1,Wait(60) exten = s,n,Playback(sorry-nobody-wants-to-let-you-in) exten = s,n,Hangup To answer just use call pickup. As long as everything is SIP you should be just fine (I think I read somewhere that IAX doesn't have call pick up). Not sure if the Local/ringforever is written quite right, but you get the idea here. Good luck, Trev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Inbound Voice Quality - Speed Change
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: ztdummy 4424 0 rtc11156 1 ztdummy zaptel178084 1 ztdummy crc_ccitt 2016 1 zaptel Ok, this is a dumb question, but what is that output from? What distribution of Linux are you using? I've never had to change anything related to the kernel. I use CentOS, though. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Inbound Voice Quality - Speed Change
Looks like output from the 'lsmod' command. Lacy Moore - Aspendora [EMAIL PROTECTED] 3/27/2007 11:34 PM On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote: ztdummy 4424 0 rtc11156 1 ztdummy zaptel178084 1 ztdummy crc_ccitt 2016 1 zaptel Ok, this is a dumb question, but what is that output from? What distribution of Linux are you using? I've never had to change anything related to the kernel. I use CentOS, though. ___ --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] ztdummy and MOH
On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. If you rub your hand across the mouthpiece of the phone, does the music play? ___ --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] cisco 7905
Khaled Chehab wrote: How to configure cisco 7905 with asterisk ,if you please can send me step by step configuration steps . This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. Sorry, can't help you because of this BS. If you want help, repost without this crap. ___ --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] ztdummy and MOH
WOW that fixed it! What an Idiot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, 28 March 2007 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ztdummy and MOH On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. If you rub your hand across the mouthpiece of the phone, does the music play? ___ --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] ztdummy and MOH
On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: WOW that fixed it! What an Idiot. I was going somewhere with that, but never mind. Good luck. Maybe the idiot is the guy who posted no additional details of his configuration, in particular, whether the CLI was showing music on hold starting, and then stopping, or whether the music on hold process was continuing but no sound. If it was a timing issue, by rubbing your hand across the mouthpiece, I would guess it is generating interupts for the timer to work and music on hold works, until you stop rubbing it and it fades it out. Hitting or tapping the mouthpiece produces the same outcome. Or, it that doesn't produce anything, it could be a permissions problem. It could be something not configured correctly in the config file. It could be that you are using mp3s instead of native format, as Andrew had asked about. But, since I'm an idiot, what do I know? ___ --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] ztdummy and MOH
The cli shows: -- Started music on hold, class 'jessica', on channel 'IAX2/205-3' -- Stopped music on hold on IAX2/205-3 I am using MP3 but I also tried it with WAV and GSM with the same result. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, 28 March 2007 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ztdummy and MOH On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: WOW that fixed it! What an Idiot. I was going somewhere with that, but never mind. Good luck. Maybe the idiot is the guy who posted no additional details of his configuration, in particular, whether the CLI was showing music on hold starting, and then stopping, or whether the music on hold process was continuing but no sound. If it was a timing issue, by rubbing your hand across the mouthpiece, I would guess it is generating interupts for the timer to work and music on hold works, until you stop rubbing it and it fades it out. Hitting or tapping the mouthpiece produces the same outcome. Or, it that doesn't produce anything, it could be a permissions problem. It could be something not configured correctly in the config file. It could be that you are using mp3s instead of native format, as Andrew had asked about. But, since I'm an idiot, what do I know? ___ --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] ztdummy and MOH
On 3/28/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: The cli shows: -- Started music on hold, class 'jessica', on channel 'IAX2/205-3' -- Stopped music on hold on IAX2/205-3 That rules out the timing. I see this note in the config file: ; If you are not using autoload in modules.conf, then you ; must ensure that the format modules for any formats you wish ; to use are loaded _before_ res_musiconhold. If you do not do ; this, res_musiconhold will skip the files it is not able to ; understand when it loads. Does that apply? Also, I'm not sure if this still applies, but at one time, you had to issue a restart command if you added any music files for the Asterisk to see them. A reload command wouldn't do it. Have you tried restart (not of the system, just Asterisk from cli). Another thing you may or not be able to check... what if you just put the files in the default directory and in the default context? Do they work then? This would eliminate some of the musiconhold config options causing problems. I guess along those lines do the default music on hold files work? ___ --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] ztdummy and MOH
Lacy, it appeared to me that he was calling himself an idiot. Thanks for some of the background on the issue, though. On 3/27/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote: On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: WOW that fixed it! What an Idiot. I was going somewhere with that, but never mind. Good luck. Maybe the idiot is the guy who posted no additional details of his configuration, in particular, whether the CLI was showing music on hold starting, and then stopping, or whether the music on hold process was continuing but no sound. If it was a timing issue, by rubbing your hand across the mouthpiece, I would guess it is generating interupts for the timer to work and music on hold works, until you stop rubbing it and it fades it out. Hitting or tapping the mouthpiece produces the same outcome. Or, it that doesn't produce anything, it could be a permissions problem. It could be something not configured correctly in the config file. It could be that you are using mp3s instead of native format, as Andrew had asked about. But, since I'm an idiot, what do I know? ___ --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] ztdummy and MOH
I am using autoload and I have rebooted the server. I have tried using different files and a different location. This is getting very frustrating. I wish I knew what the problem was. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, 28 March 2007 3:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ztdummy and MOH On 3/28/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: The cli shows: -- Started music on hold, class 'jessica', on channel 'IAX2/205-3' -- Stopped music on hold on IAX2/205-3 That rules out the timing. I see this note in the config file: ; If you are not using autoload in modules.conf, then you ; must ensure that the format modules for any formats you wish ; to use are loaded _before_ res_musiconhold. If you do not do ; this, res_musiconhold will skip the files it is not able to ; understand when it loads. Does that apply? Also, I'm not sure if this still applies, but at one time, you had to issue a restart command if you added any music files for the Asterisk to see them. A reload command wouldn't do it. Have you tried restart (not of the system, just Asterisk from cli). Another thing you may or not be able to check... what if you just put the files in the default directory and in the default context? Do they work then? This would eliminate some of the musiconhold config options causing problems. I guess along those lines do the default music on hold files work? ___ --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
