[Asterisk-Users] Asterisk hangup for no reason (FXO)
I am using asterisk as a voicemail for an old Nitsuko TIE ONYX VS system. it sends the signals through DTMF, and that works ok sometimes. but when a call comes in from outside and is transfered to voicemail or goes to voicemail because no one picks up, the following happens. the interface rings asterisk picks up waits for digits then it just "hangup" for no reason. Now comes the weird part if I split the line and just after asterisk picks up, I pick up a phone that is on the same port, I hear the digits and asterisk transfers to the right extension, and everything is good. What on earth is going on here it is driving me crazy, I am using fxs_ks for the signaling could this be doing it? Please help I am so out of ideas. Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 267.4.1 - Release Date: 6/2/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hangup on 9 key being pressed at any time
I am using asterisk as a voicemail for an old tie onyx phone system. I have almost everything working except that the stupid phone system doesn't hang uo when the user hangs it it only sends me a bunch of 9's. I put a 9 extension that hangs up and that works for the Background app, but once I get into voicemail the only thing I can do is put in an absolute timeout. can anybody give me advice on a better way to do this, because it is tying up all my lines way more than is needed. The lines are also coming in on digium fxo cards. Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 267.3.1 - Release Date: 5/31/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hum on the Sipura-841
Has anyone had problems with a small electrical type of hum on the 841's handset. It is there on all of the three phones I bought, and also do the sound like the microphone is cheap and kind of a high pitched talking into a can. I can live with these as long as I know that this is what the phones are like and I don't have duds. Thanks joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.16 - Release Date: 5/24/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX losing registration
I have them all as IP's. I tried registering with myself and that stayed up longer then the ones going out on the net, but when I did a reload then that one died too. I had this problem when I was running 1.0.5 and still have it with 1.0.7. does this function have any dependancies that I don't know about. P.S. they stup up for longer then 3 hours but sometime during the night they died. Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Riddell Sent: Sunday, May 22, 2005 11:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX losing registration Joel Duffield wrote: > The problem is still occuring. it happens even if I register with myself, it > works for some time and then just dies. The qualify still shows up as 65ms > on the outside server, but the registry just says "Request Sent". and a > reload doesn't help only restart. Are you registering against a hostname or IP? Try changing to an IP if possible. Is it maybe caching DNS somewhere along the line for that Host (and maybe the IP has changed)? Strange that reload does not work as I thought that it cleared the dns cache (as far as Asterisk is concerned). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 5/22/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 5/22/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX losing registration
The problem is still occuring. it happens even if I register with myself, it works for some time and then just dies. The qualify still shows up as 65ms on the outside server, but the registry just says "Request Sent". and a reload doesn't help only restart. Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Riddell Sent: Saturday, May 21, 2005 11:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX losing registration Joel Duffield wrote: > The firewall I'm using is a Linksys BEFSR41 V3 it says that it uses "The > router uses NAT and TCP/IP port inspections" not stateful inspections. Make sure that your are using qualify=xxx for your IAX2 peers. For example, if you set it to 400 (this is in iax.conf in the definition for a particular account), it would send a request every 400ms (and mark the peer as unreachable if it goes over this amount). If you still get problems, lower the number. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 5/22/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 5/22/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX losing registration
The firewall I'm using is a Linksys BEFSR41 V3 it says that it uses "The router uses NAT and TCP/IP port inspections" not stateful inspections. Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Francisco A. Lozano Sent: Saturday, May 21, 2005 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX losing registration Maybe you connections pass through a stateful firewall , and these states die after some inactivity time... Check it. - Original Message ----- From: "Joel Duffield" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, May 21, 2005 3:45 PM Subject: [Asterisk-Users] IAX losing registration > My * box keeps losing its registration to all the servers it is > registering > to, the only way to fix it is to restart asterisk and then it works fine > for > another 2 hours or so. I'm on a static IP, but this happens like clockwork > every time. I have seen other people that have this problem but never an > answer. Please can any guru out there help me. This is the only problem > with > this system that is keeping me from going live with it. Here is my IAX2 > show > registry. > > Host UsernamePerceived Refresh State > 216.94.102.***:4569 ** 60 Request > Sent > 139.142.184.***:4569 ** 60 Request > Sent > > * them out just for security but these fields show up fine. > > Thanks Joel > -- > No virus found in this outgoing message. > Checked by AVG Anti-Virus. > Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX-IAX Trunking not works
Okay sounds like a stupid question but just to be clear do you have some sort of timer on both machines? Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adnan Ahmed Sent: Saturday, May 21, 2005 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX-IAX Trunking not works Hello , I want some tips guidance i am sure this topic discuss alot in list,i try my best to solve it by myself try googling looking wiki everywhere but no luck question is iax-iax trunking not working setting,trying each n every option server2 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [saim] username=saim secret=saim type=friend host=dynamic context=from-sip disallow=all allow=gsm [noman] username=saim secret=noman type=friend host=dynamic context=from-sip disallow=all allow=gsm [asteriskser1] type=friend ;auth=md5 ;secret=qwerty context=local ;host=dynamic defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no server1 iax.conf: [general] bindport=4569 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay trunk=yes notransfer=yes [user1] username=user1 secret=user1 type=friend host=dynamic context=from-sip disallow=all allow=gsm [user2] username=user2 secret=user2 type=friend host=dynamic context=from-sip disallow=all allow=gsm [test2] type=friend context=local defaultip=192.168.0.51 notransfer=yes qualify=no trunk=yes canreinvite=no I am using Kiax soft phone on both servers using codec GSM asterisk latest stable version OS SLES9 ,any help is highly appreciated i had look almost every place in wiki regarding iax trunking but all in vein. Thanks In Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Confirmation Of Extension Before Transfer?
Try to use macro's I am not the one to ask about them, I couldn't give you an example off the top of my head. But read up on them on the wiki, and i'm sure they can do what you want very easily. Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Stearne Sent: Saturday, May 21, 2005 11:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Confirmation Of Extension Before Transfer? Is there any way to have the user confirm the extension they are looking to go to before transfering? i.e. "You pressed 5 4 3 3 2. Is this correct?" 1 - GoTo extensionPressed 2 - Enter extension again Thanks! Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] paging thru sipura-841
Hey steve I remember a tip somewhere where they used a conference room and added all the users into that conference muted, then kicked them out at the end of the call. Sorry I can't remember at all where this was but it looked like it could work. How did you get the autoanswer to work, I have tried different patches and non work? joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Clark Sent: Friday, May 20, 2005 9:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] paging thru sipura-841 Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX losing registration
My * box keeps losing its registration to all the servers it is registering to, the only way to fix it is to restart asterisk and then it works fine for another 2 hours or so. I'm on a static IP, but this happens like clockwork every time. I have seen other people that have this problem but never an answer. Please can any guru out there help me. This is the only problem with this system that is keeping me from going live with it. Here is my IAX2 show registry. Host UsernamePerceived Refresh State 216.94.102.***:4569 ** 60 Request Sent 139.142.184.***:4569 ** 60 Request Sent * them out just for security but these fields show up fine. Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Losing IAX registration.
I have an * 1.0.7 box that keeps loosing its registration to both of the other servers it registers with. When I start * it connects and registers fine and I can make calls. but after a few hours it shows the status as Auth. Sent but I can no longer make calls to the other servers. The other servers are not mine they are Providers servers. I am behind a nat with port forwarding, and am running on a xercom install that I upgraded to a full debian install. This has happened to me from day one. Any help would be great Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.12 - Release Date: 5/17/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Paging with the Sipura-841
I have read how to get paging working on the 841 using the SIP Header. I have tried to install chan_sip2 but the make failed and the patch that was also mentioned I cannot find to download. I am using asterisk stable 1.0.7. What is the best way to implement this with the littlest cost to other features, and to reliability? Any help would be great because I'm stuck. Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.12 - Release Date: 5/17/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with asterisk starting from init.d
Hi All I had asterisk running on a xercom install, I upgraded the box to a full debian install and now asterisk is not starting from on boot. I can start asterisk from the command line fine no problems, but when i type /etc/init.d/asterisk start it says asterisk PBX started. It doesn't start it though, when I look at the log file it has this. May 16 10:19:05 WARNING[3711]: Unable to open '/dev/zap/channel': Permission denied May 16 10:19:05 ERROR[3711]: Unable to open channel 1: Permission denied here = 0, tmp->channel = 1, channel = 1 May 16 10:19:05 ERROR[3711]: Unable to register channel '1' May 16 10:19:05 WARNING[3711]: chan_zap.so: load_module failed, returning -1 May 16 10:19:05 WARNING[3711]: Loading module chan_zap.so failed! the card works fine when I start asterisk from the command line. Can anyone help me please? Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.10 - Release Date: 5/13/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] two questions about the Sipura 841?
The beeping is to tell you that the remote end has hungup, im sorry I don't know the technical term for it but it happens on your regular home phone if the other end was to hang up and you did not hang up your receiver. the web interface calls it the "Reorder". Thanks Joel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jim Sturtevant Sent: Saturday, May 07, 2005 4:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] two questions about the Sipura 841? What is the purpose of the beeping? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Duffield Sent: Saturday, May 07, 2005 12:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] two questions about the Sipura 841? Ok my first question is I have seen messages about a patch for asterisk so that I can do auto answer on these phones. I found the message in the archives but I do not have that message as an email still, so I do not have the attachment. Can anyone tell me where to get it? Also on this phone how can I set the phone to release the line sooner without playing the anoying beeping for 5 seconds, I can change how long until the beeping starts but how do I shorten the beeps? Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] two questions about the Sipura 841?
Ok my first question is I have seen messages about a patch for asterisk so that I can do auto answer on these phones. I found the message in the archives but I do not have that message as an email still, so I do not have the attachment. Can anyone tell me where to get it? Also on this phone how can I set the phone to release the line sooner without playing the anoying beeping for 5 seconds, I can change how long until the beeping starts but how do I shorten the beeps? Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.6 - Release Date: 5/6/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dynamic phone groups.
I would think what you would need to look at is how to do this with the *
Data Base. I haven't done this, but it would seem that there is a way to
make it work with that.
Joel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert P.
McKenzie
Sent: Saturday, April 30, 2005 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dynamic phone groups.
Joris Vandalon wrote:
> Hi,
>
> I am looking for a way to dynamicly put phones in a group so if someone
> calls an extentions everyone's phone who's member of the group will
> ring.
> Queues are not an options because as soon a call comes in to a queue
> there is no getting out.
> I want to let the phones ring and after a period of time stop trying and
> continue to voicemail for example.
> Can someone provide me with some hints or examples getting this done?
It may not be exactly what you are after but I do something like this:
extensions.conf
HOUSEPHONES=SIP/somepc&SIP/anotherpc&IAX2/desktop&IAX2/someotherdesktop&SIP/
sipuraline1&SIP/sipuraline2
; London Number - SIP Inbound provider
exten => 1438645,1,Answer
exten => 1438645,2,Dial(${HOUSEPHONES}|60|t)
exten => 1438645,3,Voicemail(u50)
Each phone listed above also has it's own extention, but the voicemail all
goes to 50. That way I can call any
extension from anyother inside the house. But calling 50 directly will make
every phone ring. Any phone not logged in
will just be ignored and skipped. The first phone to pick up gets it. Call
parking is on so if there is a need to
transfer calls from one phone to another it can be done using parking.
--
Robert P. McKenzie | GammaRay Technical Services Ltd
[EMAIL PROTECTED] | [EMAIL PROTECTED]
http://www.uk-experience.com | http://www.gammaray-tech.com
Ecademy Profile: http://www.ecademy.com/account.php?op=view&id=64014
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Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.0 - Release Date: 4/29/2005
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RE: [Asterisk-Users] Re: Voicemail name (greet.wav) is not played
ï This file is used for the Directory Application. I don't think it is ever used in voicemail, it's only used to play the name before Directory forwards them to that extension. Joel -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of bamSent: Thursday, April 14, 2005 1:46 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Re: Voicemail name (greet.wav) is not playedSorry, I've not bee clear enough, when does the greet file get played at all? I can see how to record the greet.* sound file, and the documentation for that, but so far can only see the busy and unavail messages being played. There are no error messages assocated with this, just users asking what happened to the name that they recorded in response to the prompts.many thanks,Brian>On Thu Apr 14 12:22:02 CDT 2005, Roderick A. Anderson wrote:>>On Thu, 2005-04-14 at 18:05, bam wrote: How or when is the voicemail name actually played?I've recorded my name message and can see that the voicemail directory now has two new greet files and the original greet.gsm has been overwritten.# ls /var/spool/asterisk/voicemail/default/4100/INBOX/ -l -rw-r--r-- 1 root root 8943 Feb 10 17:22 busy.gsm -rw-r--r-- 1 root root 3993 Apr 14 17:31 greet.gsm -rwx-- 1 root root 38764 Apr 14 17:31 greet.wav -rwx-- 1 root root 3960 Apr 14 17:31 greet.WAV drwxr-xr-x 2 root root 4096 Feb 24 12:15 INBOX -rw-r--r-- 1 root root 8943 Feb 10 17:22 unavail.gsmThere is no mention in Wiki or Google and I've even resorted to scouring the source code, but all I can find are the options to record the name. How do I use the name option? >Could this be a permissions issue? Should greet.(wav,WAV) be the same >as greet.gsm? > > >Rod >-- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Choppy music on hold
When I put a caller on hold who is connected via our IAX provider the MOH is dreadfull, not acceptable at all. it come on for one second then off for a few. The calls sound fine when I am taking to someone, and the MOH works on the network, I also have a dummy timer installed. Has anyone had this problem before? Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.8 - Release Date: 4/13/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clipcomm CG-410 with asterisk?
has anyone used the Clipcomm 4 port FXO Gateway with asterisk? and has it worked just as a gateway should and pass calls straight to asterisk, and be very easy to place outgoing calls on? This is one of the cheapest gateways I have seen and I need to have the ports out of my asterisk box so that I can have two redundant servers using it. Thanks Joel -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.8 - Release Date: 4/13/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error cannot record voicemail
I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error opening text file for o utput -- Recording the message Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v oicemail/default/300/INBOX/msg.WAV: No such file or directory -- x=0, open writing: /var/spool/asterisk/voicemail/default/300/INBOX/msg format: wav49, (n il) Mar 24 12:48:35 WARNING[344081]: app.c:701 ast_play_and_record: Error creating writestream '/var/spo ol/asterisk/voicemail/default/300/INBOX/msg', format 'wav49' Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:784 base_encode: Failed to open log file: /var/spoo l/asterisk/voicemail/default/300/INBOX/msg.WAV: No such file or directory Does anyone know what is up with this apache isn't trying to share this anymore so did it change some permissions? Thanks Joel Duffield -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.1 - Release Date: 3/23/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] calling an extension after a voicemail is left
Hi All I am setting up * for use as a voicemail. I have discovered that if I dial the phone system and send "#+Extension+messagenumber" (dtmf) that the "msg" light will come on on the phones, if 00 messages the light will go off. they are an old tie onyx vs system. So how can I get asterisk to pick up a line and send these digits after a voicemail has been left, even if the person hangs up the phone? would I need a script to do this, I'm a noobie at writing scripts? Any experience or advice would be greatly appreciated. Thanks Joel Duffield ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best gateway to use for *?
Hi All I am working on setting up a * system to replace our current voicemail box. I may also end up using it for a few Voip calls. Anyway, I have heard some people complaining about the new Digium Fxo cards and having problems with them. I do not yet have the computer so if certain issues are caused by other hardware I could work around them. Does anyone trust these cards enough to use them? I hant to make sure this setup is clean so I can convince my boss to move to asterisk for the whole system. If these cards are no good, what other hardware is out there in the same kind of price range ($300-400) that can handle 4 Fxo's. Thanks Joel Duffield ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing a remote phone system and then entering an extension
I am trying to get a way to have * forward calls that are dialed to an extension, to end up at an extension on my old analog phone system. I will have 7 lines coming into * using the new Digium cards via PSTN, and then lines coming from * into the PSTN lines on the analog system. So that if for example someone dials extension 110: The system will call the analog system, the system will assume that a call is coming from the telco as always, pick up right away, and then listen for an extension to be entered. This should then connect the incoming call to the extension on the analog system. My question is, does my logic work, and also if I use the dial command, and I set the analog system to pick up immediately, will wait long enough before it dials? If that wouldn't work is there a way that I can tell * to dial then wait and then send digits? Thanks Joel Duffield Near North Business Machines www.NearNorthBusiness.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interfacing with an existing phone system
On Thursday 22 April 2004 07:05 pm, Joel Duffield wrote: > We want to use asterisk to extend our current phone system. It is a > regular plain old system. Has anyone done this before? Absolutely - in a lot of different ways. > We would be > adding about 4 SIP (probably Cisco) phones to use with asterisk. What > kind of card will I need to use for this, FXS or FXO. Neither of those types of cards. You will need an ethernet card/connection to use a SIP phone. Also, before you pay all that money for new phones, you could test/learn using asterisk with free softphones. - Sorry I didn't ask this question very well, I meant how will I interface with the existing phone system, It is an old system so really the only way I have to connect to it is through putting asterisk in place of a phone at an extension. Can I use the four port card? The whole thing is a trial to convince the powers that be to switch the whole system over to VOIP as the old system is on its last legs. **** Anon Thanks Joel Duffield Near North Business Machines 705-787-0517 Phone 705-787-0554 Fax [EMAIL PROTECTED] www.NearNorthBusiness.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Interfacing with an existing phone system
We want to use asterisk to extend our current phone system. It is a regular plain old system. Has anyone done this before? We would be adding about 4 SIP (probably Cisco) phones to use with asterisk. What kind of card will I need to use for this, FXS or FXO. Also does anyone have any ideas what the best way to go about this is, should I just forward existing lines to specific phones (just to save on running new telephone cabling) or would there be any simple ways to make a small menu and just put one more layer before they get through? Thanks Joel Duffield Near North Business Machines www.NearNorthBusiness.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
