RE: [Asterisk-Users] Transfering from a device to a queue crashesAsterisk
Hi David, I've got probably the same/a similar problem. Do you add the phones to the queue (AgentLogin/AddQueueMember)? If there are entries like: " Spawn extension (macro-dialout-trunk,s,21) exited non-zero..." in your * log file you might have the same problem like me. I suspect that something goes wrong with the nested macro calls within the AMP-generated dialplan, so what I did was to expand macro-dial etc.. for each local SIP extension. This seems to work, but is not really nice because you will have to configure the extensions manually (which makes AMP more or less obsolete)... cheers Jörg From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David F. BakkerSent: Thursday, September 15, 2005 5:28 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Transfering from a device to a queue crashesAsterisk Asterisk crashes with no errors when I transfer from a device (my phone) to a queue Asterisk crashes with no errors. Also if I xfer from a sip device to another and dont wait for the other user to pickup before xfering the call gets dropped. Any ideas? Im using the latest cvs of asterisk, amp 1.10.009 and our phones are polycom 501. Yahoo! for GoodClick here to donate to the Hurricane Katrina relief effort. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Help on RealTime Extensions on Oracle DB
Does someone here configured RealTime Extensions using ODBC connecting to Oracle DB? Im having a problem in dialplan patterns, it doesnt work. Pls. help! -Chris __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Call recording between SIP phones
Lakmal wrote: / I have been searching around for days on how to record calls between SIP/ / Phones. Could someone tell me whether it is possible? The Record command/ / doesn't seem to work during a call./ If you are using a fairly recent version of HEAD, you can check out Dial options w or W and features.conf. This will enable a feature code for you to press while you are on the call to start the recording. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Call recording between SIP phones
Hi everybody,> > I have been searching around for days on how to record calls between SIP> Phones. Could someone tell me whether it is possible? The Record command> doesn't seem to work during a call.> > Thanks > Ishanka -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SIP reinvite asterisk and NAT
IAX will use less than individual SIP calls when trunked, yes. I'm not sure it's a significant savings with the number of streams we may be talking about for your particular scenario, but for larger carrier trunking scenarios, it could be quite significant. On your original question, if you want to do "canreinvite=yes" for the phones, you will most certainly need to make sure that those phones are on static IPs and aren't NAT'd from each other or the * server. This raises new security questions as well along the lines of making sure that only the IPs you're expecting to get traffic from to those IPs are actually able to reach those IPs, but you might solve this problem by using internal IP addressing if you've already got a VPN link established between the site that has * now and the remote office where you're looking to bring up these phones. UDP based VPNs are better than TCP based VPNs for VoIP within VPN from a latency perspective, but I've deployed plenty of VPNs that weren't UDP based that were able to sponsor VoIP traffic through them without any problem. On 9/16/05, Jason Walker <[EMAIL PROTECTED]> wrote: I am curious...are you saying to use SIP locally and IAX from point to point(over a WAN or VPN tunnel)? With that in mind, do you think that using a lesser compressed codec over the IAX trunk would give an okay amount ofbandwidth savings?Thanks.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Mark PhillipsSent: Thursday, September 15, 2005 7:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP reinvite asterisk and NATIf these phones are all to be in a single location I'd deploy a remoteAsterisk box and run an IAX trunk between remote and local sites.That'll save more bandwidth than having a potential 5 individual SIP sessions running over your link.Also, with the addition of an analogue card such as the TDM400 series you'llhave survivability should your link go down.If you don't add a phone line to the remote site how will they be able to call 911 etc?MarkDamon Estep wrote:> I would like to setup up a remote office with a half dozen or so SIP> phones connected to an asterisk server via a WAN link. To conserve> bandwidth I would like the phones to be able to re-invite when they > call each other. The phones will be Polycom, Cisco, or Snom. I may or may not use NAT. Seems like the NAT would really mess up> re-invites, any experience with that? Assuming no NAT, what should be expected in this setup? I know the transfer option in asterisk would not work, but I do not> think that is a big deal since any re-invited calls would be user to > user, with little or no need to transfer. As long as the SIP termination peers I am using are set to> canreinvite=no then a call between the users and a remote party would > not be re-invited, since the peer terminating the call is set to no,> correct? Can someone share some experiences wit this type of setup? Are there> other real issues to look out for or be aware of? I am really just trying to avoid having another asterisk box in the> remote site to maintain, but do not want to waste bandwidth on calls> going across the office.> >>> Thanks for taking the time to share your wisdom.> --> -- >> ___> --Bandwidth and Colocation sponsored by Easynews.com -->> 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--Mark, G7LTT/KC2ENIRandolph, NJhttp://www.g7ltt.com ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Changing the sip port in sip.conf does not work
I can change the sip port to any number, and when I unload and reload chan_sip.so, I always get == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 64.1.16.172:5060 == Using TOS bits 4 == Parsing '/etc/asterisk/sip_notify.conf': Found == Registered application 'SIPDtmfMode' Is there any other way to make the port change work? Also I never got an answer about how to prevent unregistered sip phones from sending inbound SIP calls. I can send calls regardless if my softphone is registered or not, when autocreatepeer=no. This is flaw that makes Asterisk very insecure. F.A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Send SIP NOTIFY frequency
Am I missing a parameter somewhere or does a frequency (eg 5 mins) for sending NOTIFY when there is voicemail does not exist? The clients are registered on SER so if they are offline when the voicemail comes in, they miss the notify message with no chance of getting it again except if I reload Asterisk. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SIP reinvite asterisk and NAT
I am curious...are you saying to use SIP locally and IAX from point to point (over a WAN or VPN tunnel)? With that in mind, do you think that using a lesser compressed codec over the IAX trunk would give an okay amount of bandwidth savings? Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Thursday, September 15, 2005 7:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP reinvite asterisk and NAT If these phones are all to be in a single location I'd deploy a remote Asterisk box and run an IAX trunk between remote and local sites. That'll save more bandwidth than having a potential 5 individual SIP sessions running over your link. Also, with the addition of an analogue card such as the TDM400 series you'll have survivability should your link go down. If you don't add a phone line to the remote site how will they be able to call 911 etc? Mark Damon Estep wrote: > I would like to setup up a remote office with a half dozen or so SIP > phones connected to an asterisk server via a WAN link. To conserve > bandwidth I would like the phones to be able to re-invite when they > call each other. > > > > The phones will be Polycom, Cisco, or Snom. > > > > I may or may not use NAT. Seems like the NAT would really mess up > re-invites, any experience with that? > > > > Assuming no NAT, what should be expected in this setup? > > > > I know the transfer option in asterisk would not work, but I do not > think that is a big deal since any re-invited calls would be user to > user, with little or no need to transfer. > > > > As long as the SIP termination peers I am using are set to > canreinvite=no then a call between the users and a remote party would > not be re-invited, since the peer terminating the call is set to no, > correct? > > > > Can someone share some experiences wit this type of setup? Are there > other real issues to look out for or be aware of? > > > > I am really just trying to avoid having another asterisk box in the > remote site to maintain, but do not want to waste bandwidth on calls > going across the office. > > > > Thanks for taking the time to share your wisdom. > > > > > > > > > -- > -- > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Is digium supporting new te405p and te406pinstall?
I was happy with FC3, old te405p and * 1.0.7. I've been thinking that kernel 2.6 is more stable and secure. --- Jason Walker <[EMAIL PROTECTED]> wrote: > I kept running into compile errors when dealing with > my Compaq (it is an > older quad 700 Xeon...not sure of the model number). > Once I dropped to FC1, > the install of 1.0.9 compiled and install without an > issue. > > Is there some other process/app that you are running > that requires the newer > kernel? > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On > Behalf Of Jason Kim > Sent: Thursday, September 15, 2005 8:54 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Is digium supporting > new te405p and > te406pinstall? > > I tried both 1.0.9 and 1.2beta. > I couldn't see any interrupt from /proc/interrupt. > My email server has no spam filter. > > --- Jason Walker <[EMAIL PROTECTED]> wrote: > > > I have not been able to get * 1.0.9 on a FC4 > box...I have an older IBM > > server just waiting and try it every so often. > When I am using a card > > for timing (TE405P is what we pretty much use), I > feel pretty > > comfortable with > > FC1 and 1.0.9. > > > > Are you using 1.0.9? Have you tried 1.2 beta? > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] > On Behalf Of Jason > > Kim > > Sent: Thursday, September 15, 2005 7:59 PM > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] Is digium supporting new > te405p and te406p > > install? > > > > Hi, > > > > I tried to install these cards using FC3 and FC4 > on various > > motherbords, but to fail. > > I sent email to digium several times, but no > response. > > I think these cards are not for production use > yet. > > > > Regards, > > Jason > > > > __ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam > protection around > > http://mail.yahoo.com > ___ > > --Bandwidth and Colocation sponsored by > Easynews.com > > -- > > > > 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 > > > > ___ > > --Bandwidth and Colocation sponsored by > Easynews.com > > -- > > > > 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 > > > > > > > __ > Yahoo! Mail - PC Magazine Editors' Choice 2005 > http://mail.yahoo.com > ___ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > 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 > > ___ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > 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 > __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] QUESTION: RINGING CONTINUES DURING CALL
After searching around, I've been unable to to find any relevant info on this. Perhaps the group can help? I am seeing something strange with a new Sipura SPA-3000 (and I've noticed this also with an IAX softphone): When I dial 777, this dialplan (in extensions.conf) is run: exten => 777,1,Dial(Zap/1/2345678) exten => 777,n,Hangup The number is answered by the called party, but the ringing sound continues and is heard over top of the conversation. If I add an Answer line to the dialplan, this problem disappears: exten => 777,1,Answer exten => 777,n,Dial(Zap/1/2345678) exten => 777,n,Hangup This does not occur with a Grandstream BT-101 or an XTen SIP softphone. Why do some devices need the "Answer" line in the dialplan before the "Dial" line? (I'd rather not have to do a custom section for certain devices). Thanks for any info... Alvin ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Is digium supporting new te405p and te406pinstall?
I kept running into compile errors when dealing with my Compaq (it is an older quad 700 Xeon...not sure of the model number). Once I dropped to FC1, the install of 1.0.9 compiled and install without an issue. Is there some other process/app that you are running that requires the newer kernel? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Thursday, September 15, 2005 8:54 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Is digium supporting new te405p and te406pinstall? I tried both 1.0.9 and 1.2beta. I couldn't see any interrupt from /proc/interrupt. My email server has no spam filter. --- Jason Walker <[EMAIL PROTECTED]> wrote: > I have not been able to get * 1.0.9 on a FC4 box...I have an older IBM > server just waiting and try it every so often. When I am using a card > for timing (TE405P is what we pretty much use), I feel pretty > comfortable with > FC1 and 1.0.9. > > Are you using 1.0.9? Have you tried 1.2 beta? > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Jason > Kim > Sent: Thursday, September 15, 2005 7:59 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Is digium supporting new te405p and te406p > install? > > Hi, > > I tried to install these cards using FC3 and FC4 on various > motherbords, but to fail. > I sent email to digium several times, but no response. > I think these cards are not for production use yet. > > Regards, > Jason > > __ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com ___ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > 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 > > ___ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > 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 > __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Is digium supporting new te405p and te406p install?
I tried both 1.0.9 and 1.2beta. I couldn't see any interrupt from /proc/interrupt. My email server has no spam filter. --- Jason Walker <[EMAIL PROTECTED]> wrote: > I have not been able to get * 1.0.9 on a FC4 box...I > have an older IBM > server just waiting and try it every so often. When > I am using a card for > timing (TE405P is what we pretty much use), I feel > pretty comfortable with > FC1 and 1.0.9. > > Are you using 1.0.9? Have you tried 1.2 beta? > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On > Behalf Of Jason Kim > Sent: Thursday, September 15, 2005 7:59 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Is digium supporting new > te405p and te406p > install? > > Hi, > > I tried to install these cards using FC3 and FC4 on > various motherbords, but > to fail. > I sent email to digium several times, but no > response. > I think these cards are not for production use yet. > > Regards, > Jason > > __ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam > protection around > http://mail.yahoo.com > ___ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > 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 > > ___ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > 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 > __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sip recording
Hi everybody, I have been searching around for days on how to record calls between SIP Doesn’t seem to work during a call. Thanks Ishanka -- This e-mail and any attachments are intended for the above named recipient(s) only and may be privileged. This message and any attachments has been scanned for viruses and dangerous content by ITABS Lanka Mail Scanner, and is believed to be clean. Although measures have been taken to ensure that this e-mail and attachments are free from any virus we advise that, in keeping with good computing practice, the recipient should ensure they are actually virus free. Please note that this message has been sent over public networks which may not be a 100% secure communications medium and ITABS Lanka cannot be held responsible for its integrity. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Is digium supporting new te405p and te406p install?
> I tried to install these cards using FC3 and FC4 on > various motherbords, but to fail. > I sent email to digium several times, but no response. > I think these cards are not for production use yet. Have you checked your SPAM folder? My email correspondence with Digium is always prompt. The TE405/406 have worked well for me in production for almost two years now (the 405 anyway, the 406 just came out but is working well.) -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Is digium supporting new te405p and te406p install?
I have not been able to get * 1.0.9 on a FC4 box...I have an older IBM server just waiting and try it every so often. When I am using a card for timing (TE405P is what we pretty much use), I feel pretty comfortable with FC1 and 1.0.9. Are you using 1.0.9? Have you tried 1.2 beta? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim Sent: Thursday, September 15, 2005 7:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Is digium supporting new te405p and te406p install? Hi, I tried to install these cards using FC3 and FC4 on various motherbords, but to fail. I sent email to digium several times, but no response. I think these cards are not for production use yet. Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SIP reinvite asterisk and NAT
> If these phones are all to be in a single location I'd deploy a remote > Asterisk box and run an IAX trunk between remote and local sites. > That'll save more bandwidth than having a potential 5 individual SIP > sessions running over your link. One more potential point of failure, rather not. > > Also, with the addition of an analogue card such as the TDM400 series > you'll have survivability should your link go down. That's what cell phones are for. Each users extensions fails over to an individual cell number in link down situations and just for convenience. > > If you don't add a phone line to the remote site how will they be able > to call 911 etc? We have a 911 solution called PS/ALI that allows us to update the ALI information for each DID (ANI) and we route the 911 calls to a selective router form the main site. Link down - use the cell phone. > > Mark > > Damon Estep wrote: > > I would like to setup up a remote office with a half dozen or so SIP > > phones connected to an asterisk server via a WAN link. To conserve > > bandwidth I would like the phones to be able to re-invite when they call > > each other. > > > > > > > > The phones will be Polycom, Cisco, or Snom. > > > > > > > > I may or may not use NAT. Seems like the NAT would really mess up > > re-invites, any experience with that? > > > > > > > > Assuming no NAT, what should be expected in this setup? > > > > > > > > I know the transfer option in asterisk would not work, but I do not > > think that is a big deal since any re-invited calls would be user to > > user, with little or no need to transfer. > > > > > > > > As long as the SIP termination peers I am using are set to > > canreinvite=no then a call between the users and a remote party would > > not be re-invited, since the peer terminating the call is set to no, > > correct? > > > > > > > > Can someone share some experiences wit this type of setup? Are there > > other real issues to look out for or be aware of? > > > > > > > > I am really just trying to avoid having another asterisk box in the > > remote site to maintain, but do not want to waste bandwidth on calls > > going across the office. > > > > > > > > Thanks for taking the time to share your wisdom. > > > > > > > > > > > > > > > > > > > > > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > 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 > > -- > > Mark, G7LTT/KC2ENI > Randolph, NJ > http://www.g7ltt.com > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Is digium supporting new te405p and te406p install?
Hi, I tried to install these cards using FC3 and FC4 on various motherbords, but to fail. I sent email to digium several times, but no response. I think these cards are not for production use yet. Regards, Jason __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SIP reinvite asterisk and NAT
If these phones are all to be in a single location I'd deploy a remote Asterisk box and run an IAX trunk between remote and local sites. That'll save more bandwidth than having a potential 5 individual SIP sessions running over your link. Also, with the addition of an analogue card such as the TDM400 series you'll have survivability should your link go down. If you don't add a phone line to the remote site how will they be able to call 911 etc? Mark Damon Estep wrote: I would like to setup up a remote office with a half dozen or so SIP phones connected to an asterisk server via a WAN link. To conserve bandwidth I would like the phones to be able to re-invite when they call each other. The phones will be Polycom, Cisco, or Snom. I may or may not use NAT. Seems like the NAT would really mess up re-invites, any experience with that? Assuming no NAT, what should be expected in this setup? I know the transfer option in asterisk would not work, but I do not think that is a big deal since any re-invited calls would be user to user, with little or no need to transfer. As long as the SIP termination peers I am using are set to canreinvite=no then a call between the users and a remote party would not be re-invited, since the peer terminating the call is set to no, correct? Can someone share some experiences wit this type of setup? Are there other real issues to look out for or be aware of? I am really just trying to avoid having another asterisk box in the remote site to maintain, but do not want to waste bandwidth on calls going across the office. Thanks for taking the time to share your wisdom. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] triggering automatic dial-outs with Zap interface
Hello,
It seems that i spent days and days trying to make
Asterisk do automatic dial-outs and i am clueless as i
have tried everything i could.
I would like Asterisk to automatically dial-out a
specific number and leave a goodbye message.
However, the outbound call isn't triggered.
Even after i reload asterisk and restart it many
times.
I am using a TDM400 card from Digium with 4 Fxo ports
and i have Debian Linux.
Here is my dialplan in extensions.conf
[outbound]
exten=>5141234567,1,Dial(Zap/1/${EXTEN})
exten=>5141234567,2,Playback(vm-goodbye)
exten=>t,1,Hangup
=-=-=
In /var/spool/asterisk/outgoing/1.call , the content
of the file is:
Channel: Zap/1/5141234567
MaxRetries: 3
RetryTime:300
WaitTme: 20
Context: outbound
Extenson: 5141234567
Priority: 1
-=-=-=
And the zap configurations in zapata.conf are:
signalling=fxs_ks
context=outbound
channel=>1-4
Your contribution will be highly appreciated.
Kind Regards,
Mahmoud
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[Asterisk-Users] Asterisk and Zyxel Prestige 2000W_v2
Hi Has anybody tried to use the Zyxel Prestige 2000W_v2 with Asterisk? I have the latest firmware and can now make outbound calls. So it looks like Asterisk does accept the configuration in sip.conf. However, I cannot receive any calls. Also, I keep getting this message on the console: chan_sip.c:7733 handle_request: Registration from '' failed for '10.99.1.151'. TIA Klaus ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo on SPA-3000 FXO
On Wed, September 14, 2005 2:42 pm, Paul Dugas wrote: > After rebooting the SPA-3000, the internal users of calls routed through > the FXO interface hear pretty bad echo. This persists for days, maybe > more than a week. At some point, the echo goes away. I've noticed that, > when the echo is gone, I hear a rapid series of light clicks on the line > when placing a call; after dialing and before the remote end starts > ringing. When the *is* echo, I'm not hearing the clicks. Per a suggestion, I swapped out the offending unit with a brand-new unit that came running 2.0.13(GWa) firmware. It's worked fine for the few test calls I just made. Will leave that in place for a couple days and see. If it stays working, I'll want to downgrade the firmware in the original unit. Is it safe to downgrade them? Thanks for all the suggestions, Paul -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC [EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Park http://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA -- Onsite at GDOT W.Annex 404-463-2860 x199 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] USB Phones for use with Asterisk
Hi all, I have a question that I was hoping someone could answer for me. I would like to find a USB phone that works with Asterisk... preferably one that does not have any issues with NAT. RESPONSE: This is Bill McCready from PCPhoneline.com . I suggest you give our VTA1000 Skype+SIP+IAX2 Gateway which is USB based with a good cordless phone. We offer several versions of the VTA1000 but this is the latest one we have developed and it supports Skype + 10 x SIP + 10 x IAX2 networks all at the same time and plugs into the USB port of a Windows 2000 or Windows XP computer. You can optionally disable Skype and/or the SIP calling capabilities if you don't need them. The VTA1000 is designed to work even if your computer screen is powered off using it with a cordless phone of course gives you the added advantage of not needing to be tied to your computer to place and receive calls. If you don't mind being tethered to a computer we also have the VPT1000 phone available with the same capability set. You can find them in our Ebay store at http://stores.ebay.com/pcphoneline . If you need a bunch of these and want a build specially tailored and branded please private e-mail me with your requirements and we will do our best to try to accomodate you. Best regards...Bill Can anyone point me to something suitable ? We are essentially planning to do a beta for a few hundred customers on our network and would like them to be able to call one another @ no charge. Because these are going to be end users who have no idea about Voice / VOIP stuff... it needs to be extremely easy to use. Can someone point me in the right direction ? Cheers, Callum ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SIP reinvite asterisk and NAT
I would like to setup up a remote office with a half dozen or so SIP phones connected to an asterisk server via a WAN link. To conserve bandwidth I would like the phones to be able to re-invite when they call each other. The phones will be Polycom, Cisco, or Snom. I may or may not use NAT. Seems like the NAT would really mess up re-invites, any experience with that? Assuming no NAT, what should be expected in this setup? I know the transfer option in asterisk would not work, but I do not think that is a big deal since any re-invited calls would be user to user, with little or no need to transfer. As long as the SIP termination peers I am using are set to canreinvite=no then a call between the users and a remote party would not be re-invited, since the peer terminating the call is set to no, correct? Can someone share some experiences wit this type of setup? Are there other real issues to look out for or be aware of? I am really just trying to avoid having another asterisk box in the remote site to maintain, but do not want to waste bandwidth on calls going across the office. Thanks for taking the time to share your wisdom. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] USB Phones for use with Asterisk
Callum McGillivray wrote: > Hi all, > > I have a question that I was hoping someone could answer for me. > > I would like to find a USB phone that works with Asterisk... preferably > one that does not have any issues with NAT. > > Can anyone point me to something suitable ? We are essentially planning > to do a beta for a few hundred customers on our network and would like > them to be able to call one another @ no charge. > > Because these are going to be end users who have no idea about Voice / > VOIP stuff... it needs to be extremely easy to use. > > Can someone point me in the right direction ? My Asia/Pacific distributor should contact you shortly. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Faxibility in NZ
Simon wrote: > Hi There, > > I understand that asterisk can recieve faxes, but im wondering if anyone > has got it to work with telecom faxibility in NZ? e.g. get asterisk to > ignore the call if it rings with the faxibility ring? Have a search for distinctive ring. Asterisk can recognize different rings. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Still having hangup problems in NZ
> Any other suggestions please??! Send me a copy of a wave file with the recorded beeps after hangup and I will see if the tones are somehow different. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] ZyXEL P662HW / SIP / Crashing
Set Wild guess mode on (I'm not familiar with zyxel).: asterisk-users wrote: Has anyone experiences this please: - We were running a number of ZyXEL P662HW-61 routers at our sites and all traffic was being sent over IP-SEC VPN's between devices. When we moved to a new architecture, we got rid of the VPN links so that the SIP traffic was running directly through the routers. Each site uses Snom 360 devices with the latest firmware (v4). This means SIP/call signalling and RTP traffic, I assume. Have you pinpointed which traffic is the culprit ? (maybe with routing for the SIP proxy via one router, and for the media gateways end points via another ? or SIP via tunnel, rtp outside ?) Ever since we did this, the routers have been crashing at least 5 times per day. They appear to carry out a full cold start each time (as though they are having a kernel panic). The ISP is Nildram in the UK, but we have also experienced this a few times with another router in France on a France Telecom system. As soon as we route the SIP traffic via another router, stability returns to the network. Our supplier has been very helpful and we have tested every release of the firmware from the last 8 months, but they all behave the same once SIP is being transmitted. The routers are running with their most basic configuration now, but this doesn't appear to make any difference. Seems like these boxes can do some level of SIP "ALG", looking/messing around with the SIP header. Can you disable this ? You depend in the SIP ALG to handle NAT or something like that ? Does this sound familiar to anyone please? We are out our wits end and our supplier has no ideas (and neither do ZyXEL it would appear). For reference, all traffic is being sent through using G.729, but I don't think that this makes any difference. If you identify what is the traffic (RTP or SIP), next step would be to identify what is the exact trigger (like, if is SIP, is a memory leak in INVITEs passing via the ALG and goes on) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] ZyXEL P662HW / SIP / Crashing
Has anyone experiences this please: - We were running a number of ZyXEL P662HW-61 routers at our sites and all traffic was being sent over IP-SEC VPN's between devices. When we moved to a new architecture, we got rid of the VPN links so that the SIP traffic was running directly through the routers. Each site uses Snom 360 devices with the latest firmware (v4). Ever since we did this, the routers have been crashing at least 5 times per day. They appear to carry out a full cold start each time (as though they are having a kernel panic). The ISP is Nildram in the UK, but we have also experienced this a few times with another router in France on a France Telecom system. As soon as we route the SIP traffic via another router, stability returns to the network. Our supplier has been very helpful and we have tested every release of the firmware from the last 8 months, but they all behave the same once SIP is being transmitted. The routers are running with their most basic configuration now, but this doesn't appear to make any difference. Does this sound familiar to anyone please? We are out our wits end and our supplier has no ideas (and neither do ZyXEL it would appear). For reference, all traffic is being sent through using G.729, but I don't think that this makes any difference. Many thanks John ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] USB Phones for use with Asterisk
Hi all, I have a question that I was hoping someone could answer for me. I would like to find a USB phone that works with Asterisk... preferably one that does not have any issues with NAT. Can anyone point me to something suitable ? We are essentially planning to do a beta for a few hundred customers on our network and would like them to be able to call one another @ no charge. Because these are going to be end users who have no idea about Voice / VOIP stuff... it needs to be extremely easy to use. Can someone point me in the right direction ? Cheers, Callum ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk CDRs
Never mind, i found a way to do this. There is an app called 'ResetCDR' which pushes the CDR info into database. Just call it with appropriate channel info and 'w' (second arg) for force insert whenever during the call processing in your dial-plan. Thanks anyway, Prakash From: Nathan Pralle <[EMAIL PROTECTED]> Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk CDRs Date: Thu, 15 Sep 2005 17:48:25 -0500 Does anyone know if there is a way to get CDR info into database before call ends??? The only way I have figured out how to interact with the call before it hangs up is to use the AGI to do various things -- whether it be inserting DB rows or whatever -- both before and after the call. Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
--On September 14, 2005 10:34:29 AM -0600 Colin Anderson <[EMAIL PROTECTED]> wrote: Disclaimer: Not a troll I'm curious as to this obsession with uptime is. All of the posts of this type are along the lines of "After X days, Y thing does not work but if I reload or reboot, it's OK" - so why not cron a reboot? Is it considered bad form or something like that? I reboot every night whether it is needed or not, not afraid to admit it, and everything works fine for me. You obviously come from a land of Windows. I have machines that are up for years at a time internally, and externally months, depending on how often security patches or other critical upgrades need applying. That's how computers should run. MS is the crap that gets people into this damned bad habit of oh i rebooted and it's fixed. No it's NOT fixed, you just got rid of the symptoms, the BUG is still there. For most UNIX and derivitaves/workalikes reboots are unnecessary. Only if something is broken, or seriously wedged. We also do the "Sunday reboot" of all of our Windows servers as well as restarting all of the critical services such as IIS , SQL, Exchange etc nightly. It helps, a lot (Exchange is a notorious memory leaker) Yup, MS you need to because MS is broken. I've also heard it said, something along the lines of: "If you have to reboot, your server isn't set up correctly" to which I say piffle. Even NASA has rebooted the Mars probes after they land and I understand that they run VXWorks, incidentally, the same RTOS that my Mitel 3300 uses, and *even Mitel* recommends periodic reboots, which we duly cron every night, 2 AM. Mitel isn't exactly known for telco grade operations. And telco is usually the one that everyone quotes as being some of the most reliable systems in the world. Imagine if Cisco or Juniper had this flawed mentality? Or Fore? Or IBM? Or Acatel? Or Nortel? Yes I probably sound like a kook or a troll but so be it. If a system is doing repeatable operations, and cleaning up correctly after itself, and doing periodic maintenance correctly, a reboot should never, under any circumstances be necessary. Even restarting the services should be a rare item (hence one of my beefs with Apache mod_ssl). 24/7/365 installs aside, is there a reason why reboots seem to be frowned upon? Again, not trolling, just curious. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk CDRs
Does anyone know if there is a way to get CDR info into database before call ends??? The only way I have figured out how to interact with the call before it hangs up is to use the AGI to do various things -- whether it be inserting DB rows or whatever -- both before and after the call. Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk CDRs
Thats because i am interested in the other fields of CDR like the Trunk/dst/application etc... Prakash From: "William Boehlke" <[EMAIL PROTECTED]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Subject: RE: [Asterisk-Users] Asterisk CDRs Date: Thu, 15 Sep 2005 15:40:15 -0700 Why would you want it to do that? If the call is not complete, the system cannot know the duration. Call duration is the heart of a call detail record. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Prakash Rao Kanthi Sent: Thursday, September 15, 2005 3:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk CDRs Hi Group, I am trying to capture CDR records from Asterisk into PGSQL database. The problem is that the CDR information does not get populated until the call is hangup. Does anyone know if there is a way to get CDR info into database before call ends??? Thanks, Prakash ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: 7.0.344 / Virus Database: 267.11.0/103 - Release Date: 9/15/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.0/103 - Release Date: 9/15/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
--On September 15, 2005 4:26:01 PM -0600 Colin Anderson <[EMAIL PROTECTED]> wrote: Agreed but again -- you designed for this specific purpose. Asterisk isn't meant to boot up, answer the phone, process the call and shut down again until the next ring. (This would be an interesting approach to power savings though if your system boot time was fast enough and call volumes varied enough to make it worthwhile.) That's a friggin cool idea. I wonder if someone on the list will be inspired enough to run with this: http://www.linuxbios.org/index.php/FAQ#What_is_LinuxBIOS.3F Current fastest boot time is 3s. You can have a flash filesystem for Asterisk, which is supported. How would ring detection work? You would need a "Wake on POTS" or something like that. Most modern motherboards with some sort of integrated AMR/Modem platform support wake on interrupt for com port or for modem activity... an old style modem, wake on irq for comhackish. :) i'm sure that someone makes a 'ring to restart' box thoguh too... ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk CDRs
Why would you want it to do that? If the call is not complete, the system cannot know the duration. Call duration is the heart of a call detail record. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Prakash Rao Kanthi Sent: Thursday, September 15, 2005 3:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk CDRs Hi Group, I am trying to capture CDR records from Asterisk into PGSQL database. The problem is that the CDR information does not get populated until the call is hangup. Does anyone know if there is a way to get CDR info into database before call ends??? Thanks, Prakash ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: 7.0.344 / Virus Database: 267.11.0/103 - Release Date: 9/15/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.0/103 - Release Date: 9/15/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk CDRs
Hi Group, I am trying to capture CDR records from Asterisk into PGSQL database. The problem is that the CDR information does not get populated until the call is hangup. Does anyone know if there is a way to get CDR info into database before call ends??? Thanks, Prakash ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
>Agreed but again -- you designed for this specific purpose. Asterisk isn't >meant to boot up, answer the phone, process the call and shut down again >until the next ring. (This would be an interesting approach to power savings >though if your system boot time was fast enough and call volumes varied >enough to make it worthwhile.) That's a friggin cool idea. I wonder if someone on the list will be inspired enough to run with this: http://www.linuxbios.org/index.php/FAQ#What_is_LinuxBIOS.3F Current fastest boot time is 3s. You can have a flash filesystem for Asterisk, which is supported. How would ring detection work? You would need a "Wake on POTS" or something like that. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Seperate Incoming calls on TDM02?
On Sep 15, 2005, at 2:14 PM, Matt Fredrickson wrote:On Thu, Sep 15, 2005 at 12:04:41PM -0400, C. Hatton Humphrey wrote: I have a TDM02B to bring in two POTS lines for my incoming calls; Ineed to point each line to a different IVR... is there somewhere thatcan I can look to get this setup working?Basically, each line is for a different business. I know that for aDID the routing is simple but I'm not seeing where I can match up aDID with a Zap channel.I'm currently looking into the zapata.conf file to do this as it is myunderstanding that the control can be taken care of there. My systemis running [EMAIL PROTECTED] 1.5. Yeah, in your zapata.conf just give each channel a different context setting.It's slightly more complicated with [EMAIL PROTECTED] and/or AMP, you need to use the zapata_custom.conf file, instead. You also need to use the extensions_custom.conf file, too, though there might be a better way I don't know about.Tom___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Caller ID for auto outgoing calls
I did this for our website (to be released RSN), it has a contact form that
the customer plugs in their phone number. When they do, Asterisk calls them
and dumps them to an IVR. Pressing 1 in the IVR takes them to a salesperson.
My working config is:
In the .call file:
CallerID: 194 (our 800 number)
SetVar: CALLINGPARTY=(caller ID of guy)
In extensions.conf:
[website-caller]
exten => 1,1,SetCallerID(Web Caller: ${CALLINGPARTY}) (I set the caller ID
that way so the salesperson knows that the caller is calling from the
website)
In your case, the variable you have set happens to be the same name as a
global Asterisk variable, so I'm sure that's the root of the problem. Set
the variable to a different name, try CALLINGPARTY like I did
hth
-Original Message-
From: Jim Gottlieb [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 15, 2005 3:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Caller ID for auto outgoing calls
Hi. I'm using /var/spool/asterisk/outgoing files to place automatic
calls, but I'm having trouble setting the Caller ID for the second half
of the call.
In other words, when we call the first number, we want the Caller ID
set to our number, but then when we connect them to the second number,
we want _their_ number to be the Caller ID.
I've tried the following (and various approximations):
Channel: Local/[EMAIL PROTECTED]
Callerid: 6193647100
MaxRetries: 5
RetryTime: 60
WaitTime: 60
Context: outtrunk
Extension: 16193647100
Priority: 1
SetVar: CALLERIDNUM=6193644799
When it calls 6193644799, it properly shows a Caller ID of 6193647100.
But then when it dials 6193647100, it still shows Caller ID of
6193647100 instead of 6193644799.
What am I doing wrong? How do I get the Caller ID set correctly for
the second half of the call? I've tried various other variables but I
haven't been able to get anything to work.
Thanks.
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RE: [Asterisk-Users] Asterisk 1.0.9 long term stability <--threadhijack, why not reboot?
"Just reboot" is a bad attitude. If there is a memory leek, the fact that a reboot will free the leaked memory is not a good reason to not fix the memory leek. That kind of attitude is why windows does need regular reboots. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Wednesday, September 14, 2005 11:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.0.9 long term stability <--threadhijack, why not reboot? Yes.. because if a reboot is needed it isn't setup correctly. Reboots are a windows things. A correctly setup Linux server should never need rebooted. On 9/14/05, Colin Anderson <[EMAIL PROTECTED]> wrote: > Disclaimer: Not a troll > > I'm curious as to this obsession with uptime is. All of the posts of this > type are along the lines of "After X days, Y thing does not work but if I > reload or reboot, it's OK" - so why not cron a reboot? Is it considered bad > form or something like that? I reboot every night whether it is needed or > not, not afraid to admit it, and everything works fine for me. > > We also do the "Sunday reboot" of all of our Windows servers as well as > restarting all of the critical services such as IIS , SQL, Exchange etc > nightly. It helps, a lot (Exchange is a notorious memory leaker) > > Of course, if your install processes calls 24/7 that's a different story. > However, I expect that the majority of Asterisk installs are for a 9-to-5 > type of operation. We run two shifts here, and we stop processing calls at > 10 PM, and start again at about 6 AM - a large window of opportunity to > reboot. Why not take advantage of it? > > I've also heard it said, something along the lines of: "If you have to > reboot, your server isn't set up correctly" to which I say piffle. Even NASA > has rebooted the Mars probes after they land and I understand that they run > VXWorks, incidentally, the same RTOS that my Mitel 3300 uses, and *even > Mitel* recommends periodic reboots, which we duly cron every night, 2 AM. > > 24/7/365 installs aside, is there a reason why reboots seem to be frowned > upon? Again, not trolling, just curious. > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Console/dsp and mplayer
I am wanting to share the Console/dsp port between asterisk and mplayer. I have alsa running on the box. In module.conf I did noload for chan_oss. (actually I tried it both ways) and if asterisk is running mplayer wont run. Any thoughts on what I might be missing to get these two programs to share the port? Individually everything is working. Jerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Caller ID for auto outgoing calls
Hi. I'm using /var/spool/asterisk/outgoing files to place automatic calls, but I'm having trouble setting the Caller ID for the second half of the call. In other words, when we call the first number, we want the Caller ID set to our number, but then when we connect them to the second number, we want _their_ number to be the Caller ID. I've tried the following (and various approximations): Channel: Local/[EMAIL PROTECTED] Callerid: 6193647100 MaxRetries: 5 RetryTime: 60 WaitTime: 60 Context: outtrunk Extension: 16193647100 Priority: 1 SetVar: CALLERIDNUM=6193644799 When it calls 6193644799, it properly shows a Caller ID of 6193647100. But then when it dials 6193647100, it still shows Caller ID of 6193647100 instead of 6193644799. What am I doing wrong? How do I get the Caller ID set correctly for the second half of the call? I've tried various other variables but I haven't been able to get anything to work. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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[2]: [Asterisk-Users] Indications for Ireland
Hello Ronan,
Thursday, September 15, 2005, 10:13:13 AM, you wrote:
> Hi Sean,
> This is what I've got in my zaptel zonedata.c file for a small * box in
> Dublin:
>{ 18, "ie", "Ireland", { 400, 200, 400, 2000 },
>{
>/* Dialtone = 400//425//450 */
>{ ZT_TONE_DIALTONE, "425" },
>{ ZT_TONE_BUSY, "425/500,0/500" },
>/* Ringtone = 400+450//425 */
>{ ZT_TONE_RINGTONE,
> "400+450/400,0/200,400+450/400,0/2000" },
>{ ZT_TONE_CALLWAIT, "425/180,0/200,425/200,0/4500" },
>{ ZT_TONE_INFO, "950/330,1400/330,1800/330,0/1000" },
>{ ZT_TONE_STUTTER, "350+440" },
>{ ZT_TONE_CONGESTION, "400/400,0/350,400/225,0/525" },
>{ ZT_TONE_DIALRECALL, "350+440" } },
>},
Strange thing that my [EMAIL PROTECTED] box doesn't seem to recognize the ie
prefix when
I use it in the /etc/zaptel.conf file
Sean
--
+---+
|VOIP= FreeWorldDial: 689482 VOIPBUSTER: thecivvie |
|GPG Key http://thecivvie.fastmail.fm/thecivvie.asc |
+---+
Strange things happen under the midnight sun
when Men and Dogs go hunting for gold
smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [Asterisk-Users] USB ISDN (OT question)
Hi Jorg, Ha ha - I tend to keep my off grid projects cheap & simple. I noticed that the gateways battery pack was 18V so I opened the battery pack up, removed the batteries and connected a cable to the power connector inside and ran it out through a hole I made in the back of the battery pack. Now I have the notebook connected to a 12V battery pack (500AH but it powered other stuff too: a WiFi access point and a switch and a 12 - 5V adapter to power a couple of budgetone phones and some data capture stuff). There is around 200W of PV panels connected connected to the battery back via a steca (german) 30A charge controller. Its interesting to note that the notebook happly runs with power down to around 10.5V and its been running like this for around 2 years now. I also got the notebook free because the guy who had it broke the LCD panel so now its a notebook less the screen (which I removed). The RE thing is a hobby of mine so I'm glad to answer any questions on it! All the best, Derek Jörg Wolf wrote: Derek, could you give me some details regarding the solar power supply you're using for your installation? Thanks! Jörg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Conniffe Sent: Thursday, September 15, 2005 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] USB ISDN In a remote location (solar powered) I have an old Gateway 333mhz notebook running Asterisk and using a PCMCIA AVM Fritz ISDN card. With the AVM CAPI it works perfectly with chan_capi. Derek Julien Goodwin wrote: (...) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Derek Conniffe Rivertower Ltd Ireland: (Freephone) 1800 719 400 Ireland: (Local) 01 244 9719 United Kingdom: 0870 068 2368 International: 00 353 1 244 9719 Derek Conniffe DDI: 01 201 0146 (International: 00 353 1 201 0146) Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823) Fax: 01 201 0085 (International: 00 353 1 201 0085) Email: [EMAIL PROTECTED] Web: http://www.rivertowerhosting.com begin:vcard fn:Derek Conniffe n:Conniffe;Derek org:Rivertower Ltd;IT adr:Dublin 2;;46 Upper Mount Street;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 201 0146 tel;fax:+353 1 201 0085 tel;cell:+353 86 856 3823 note;quoted-printable:Ireland: (Freephone) 1800 719 400=0D=0A= Ireland: (Local) 01 244 9719=0D=0A= United Kingdom: 0870 068 2368=0D=0A= International: 00 353 1 244 9719=0D=0A= url:http://www.rivertowerhosting.com version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Faxibility in NZ
Hi There, I understand that asterisk can recieve faxes, but im wondering if anyone has got it to work with telecom faxibility in NZ? e.g. get asterisk to ignore the call if it rings with the faxibility ring? Thanks Simon ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Still having hangup problems in NZ
Hi There, Thanks for all your suggestions. I have now compiled asterisk from cvs running on FD4. I have performed all the suggested configurations: > busydetect=yes ;changed 17.03.04 from no > busycount=7 ; added as above > for me the distro asterisk package didnt hang up properly on busy > > signal. I > needed to download the source and uncomment BUSYDETECT_MARTIN (afair) > in the > makefile before compilation to make it work. Please also remember to > load the > right tone zone (/etc/zapata.conf) for your country, otherwise > asterisk won't > be able to recognize the busy signal But the thing will just not hang up! Any other suggestions please??! Cheers! Simon ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] RxFax/TxFax - Compile Problem
I’ve reduced my problem down to this: [EMAIL PROTECTED]:/usr/src/asterisk/asterisk-1.0.9/apps# make cc -D_GNU_SOURCE -o app_rxfax.so app_rxfax.c -lspandsp -ltiff app_rxfax.c: In function `rxfax_exec': app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:264: error: structure has no member named `verbose' app_rxfax.c:325: warning: passing arg 1 of `fax_release' from incompatible pointer type make: *** [app_rxfax.so] Error 1 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Sampson Sent: Thursday, September 15, 2005 12:17 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RxFax/TxFax - Compile Problem I used the latest version (.3) and also the previous .2 ver (pre20). The spandsp seems to compile but when I download the rxfax/txfax .c files and drop them in the apps directory that is where I get the compile error. Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, September 14, 2005 2:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem What version of spandsp are you attempting to compile in to the 1.0.9 tree? On 9/14/05, David Sampson <[EMAIL PROTECTED]> wrote: Anyone know how to fix this? gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:302: error: for each function it appears in.) app_rxfax.c: In function `rxfax_exec': app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:264: error: structure has no member named `verbose' app_rxfax.c:325: warning: passing arg 1 of `fax_release' from incompatible pointer type make[1]: *** [app_rxfax.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Unable to call some numbers with I4L
Massimo Frisoni ha scritto: I have an EICON DIVA PCI 2.02 with I4L. I'm unable to call some numbers, in general numbers with automatic responders that do not rings. It's seems asterisk does not understand that the other party has answered, so after a timeout it reports 'busy', but in real the other end has answered. Any other call to a "normal" number works fine. You should probably try chan_capi and enable inband call progress reporting. (IIRC, that's done by putting "b" in the dialstring before the dialed number!) I wouldn't know how the same thing is done through I4L, maybe somebody else can shed light on that... Bye, -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
Andrew Kohlsmith wrote: On Thursday 15 September 2005 11:38, Paul wrote: They designed it to be shut down. I guess that means it doesn't just roll over like a dead cow. Actually dead cows aren't back-heavy. They typically just keep whatever position they were in when they took their last breath, much like the telco crosspoint switches. :-) In order to argue that point I would have to do some rather inhumane research :-D You wouldn't design a crane controller so that it releases the load on reboot and then tries to return the cable to the pre-reboot position. The same principle applies here. If it's mission-critical, you might have to spend more on the hardware. A good example would be crosspoint I agree -- You need to spend more time on both hardware and software. For me, a phone system is like a file server or network switch -- it is *not* meant to be rebooted. If it needs to be periodically power-cycled then address the problem, don't simply put a cron script in. I did hardware and software design for some devices that were usually placed in very remote areas(underwater, mountains, arctic are a few). We used a very low power clock device that would boot the cpu up. We did whatever needed to be done, set the clock registers for the next wakeup and shut down again. There were other ways to approach the problem, but this approach made coding easier, used less power and allowed us to get more functionality without increasing rom or ram size. If you only How did it save you ROM/RAM? I can see it saving having to put fancy power controller code and hardware in... Is that what you meant? The short answer is that the coding was more linear. Boot, run a state observer, setup the clock chip for next alarm and write an I/O bit that kills the power bus feeding the CPU area of the board. When data was being unloaded through the serial port, I skipped the final write and looped back to the top of the state observer. needed to take one simple measurement and store it to the ram, you could do a boot/run/shutoff every second and still achieve some additional battery runtime. Agreed but again -- you designed for this specific purpose. Asterisk isn't meant to boot up, answer the phone, process the call and shut down again until the next ring. (This would be an interesting approach to power savings though if your system boot time was fast enough and call volumes varied enough to make it worthwhile.) I have seen windows systems that boot up and crash. Would this be a good start? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] internet connection between Africa and Europe
Many of the satellite companies block voip because they have the sevice for sale them selfes. And dedicated satellite internet is VERY expensive. We arranged a 512/512 connection today for a callcenter in Nigeria and they will pay 6000 usd per month. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan de Konink Sent: den 15 september 2005 21:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] internet connection between Africa and Europe Check out google with: VSAT Africa, lots of companies provide IP links overthere. If it is good enough for voip... I don't yet know. Stefan On Thu, 15 Sep 2005, Jean-Michel Hiver wrote: > Stéphane LAVRI a écrit : > > >Hi > > > >I'm looking for a company who can provide me an Internet connection > >between africa and Europe. > > > > > 'Africa' and 'Europe' are both rather big, so what you're saying doesn't > make much sense. Pehaps if you outlined your requirements a bit better, > you could get some useful advice. > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] internet connection between Africa and Europe
Check out google with: VSAT Africa, lots of companies provide IP links overthere. If it is good enough for voip... I don't yet know. Stefan On Thu, 15 Sep 2005, Jean-Michel Hiver wrote: > Stéphane LAVRI a écrit : > > >Hi > > > >I'm looking for a company who can provide me an Internet connection > >between africa and Europe. > > > > > 'Africa' and 'Europe' are both rather big, so what you're saying doesn't > make much sense. Pehaps if you outlined your requirements a bit better, > you could get some useful advice. > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Unable to call some numbers with I4L
Hi all, I have an EICON DIVA PCI 2.02 with I4L. I'm unable to call some numbers, in general numbers with automatic responders that do not rings. It's seems asterisk does not understand that the other party has answered, so after a timeout it reports 'busy', but in real the other end has answered. Any other call to a "normal" number works fine. Did somebody experience this problem ? Thank you in advance for your help. Massimo Frisoni ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] internet connection between Africa and Europe
Stéphane LAVRI a écrit : Hi I'm looking for a company who can provide me an Internet connection between africa and Europe. 'Africa' and 'Europe' are both rather big, so what you're saying doesn't make much sense. Pehaps if you outlined your requirements a bit better, you could get some useful advice. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Can not get realtime static voicemail.conf to work
did you edit extconfig and put a line similar to voicemail.conf -> realtime,mysql,database then delete voicemail.conf from your asterisk configs directory and try again. matt Matt wrote: Here is what happens on startup: Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime: Static SQL: SELECT category, var_name, var_val, cat_metric FROM settings WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime: Everything is fine. Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime: Found 13 rows. Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: VM Review Option disabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: VM Operator break disabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: VM CID Info before msg disabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: Send Voicemail msg disabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: ENVELOPE before msg enabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: Duration info before msg enabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: We are not going to skip to the next msg after save/delete But that's not what I have defined in my voicemail database: mysql->asterisk_vm->settings ++++---++--+--+-+ | id | cat_metric | var_metric | commented | filename | category | var_name | var_val | ++++---++--+--+-+ | 1 | 0 | 0 | 0 | voicemail.conf | default | format | wav|gsm | | 2 | 0 | 0 | 0 | voicemail.conf | default | attach | yes | | 3 | 0 | 0 | 0 | voicemail.conf | default | pbxskip | yes | | 4 | 0 | 0 | 0 | voicemail.conf | default | serveremail | [EMAIL PROTECTED] | | 5 | 0 | 0 | 0 | voicemail.conf | default | fromstring | xx Digital Voicemail | | 6 | 0 | 0 | 0 | voicemail.conf | default | maxmessage | 180 | | 7 | 0 | 0 | 0 | voicemail.conf | default | minmessage | 5 | | 8 | 0 | 0 | 0 | voicemail.conf | default | maxsilence | 5 | | 9 | 0 | 0 | 0 | voicemail.conf | default | silencethreshold | 128 | | 10 | 0 | 0 | 0 | voicemail.conf | default | skipms | 3000| | 11 | 0 | 0 | 0 | voicemail.conf | default | review | yes | | 12 | 0 | 0 | 0 | voicemail.conf | default | operator | no | | 13 | 0 | 0 | 0 | voicemail.conf | default | emailbody| You have a new voicemail! | ++++---++--+--+-+ Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 sip before answering pstn line
Hello, I have asterisk server with two isdn bri cards (billion) using zaphfc driver. Also I have from telephone company routed (for example) 16 pstn numbers. It is technically possible to dial SIP phone from outside before answering isdn pstn line. I have local numbers 201,202,203 and from telecomunication company numbers 555201,555202,555203. I don't want to be "pstn" caller billed if SIP user don't answer. It is possible? thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] If call fails, then try again with something else
What is a good way to set up in the dialplan for the case where a call fails (say due to congestion or whatever) and then asterisk immediately dials again, with a different trunk or perhaps another destination number? Thanks -- Eric Smith ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Call Pickup between ZAP and SIP technologies
Hi, I have this scenario. In my desk I have a phone connected to a FXS module of my * server. On another desk there is a phone but it is a SIP softphone (SJphone). I hear the SIP softphone is ringing, then I try to take that call with my Zap phone in my desk dialing *8, but I get fast busy tone. Is there I way do this to work ? I mean pickup phones that are ringing on different technologies ? Ardg. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Seperate Incoming calls on TDM02?
On Thu, Sep 15, 2005 at 12:04:41PM -0400, C. Hatton Humphrey wrote: > I have a TDM02B to bring in two POTS lines for my incoming calls; I > need to point each line to a different IVR... is there somewhere that > can I can look to get this setup working? > > Basically, each line is for a different business. I know that for a > DID the routing is simple but I'm not seeing where I can match up a > DID with a Zap channel. > > I'm currently looking into the zapata.conf file to do this as it is my > understanding that the control can be taken care of there. My system > is running [EMAIL PROTECTED] 1.5. Yeah, in your zapata.conf just give each channel a different context setting. -- Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Phonecall or something as robust
Joshua Abbott wrote: Has anyone every heard of Phonecall? : www.vecsector.com/phonecall/ Feedback? Is there something as good as it or better ? Recommendations? I've heard of it! ;-) Currently, the biggest trouble with it is the hardware configuration. I'm working on a new Hardware Manager to help resolve these issues. We've had a few bugs that have been worked out - thanks to everyone who submitted them on http://bugs.vecsector.com I've had a few programmers offer to help write a plugin to FOP that will autogenerate the configurations (op_buttons.cfg), realtime support, meetme control & a queue manager for the next version 2.7. I'm sure it's not perfect, but I think with more feedback like you are requesting, we can really iron this out to be a nice complimentary GUI for Asterisk. Feedback is most welcome, either onlist or offlist. Thanks! --Dustin Wildes ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Seperate Incoming calls on TDM02?
There may be a better way, but you can use the incoming call settings in asterisk to point the first line at the first IVR. Then, if you search the archives, you will find a post by me regarding custom incoming routing and AMP, which describes how I did it. Be sure to read the whole thread, because my initial post had some errors in it. Tom On Sep 15, 2005, at 12:04 PM, C. Hatton Humphrey wrote: I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file to do this as it is my understanding that the control can be taken care of there. My system is running [EMAIL PROTECTED] 1.5. Help! Hatton ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] internet connection between Africa and Europe
FlagTel offer dedicated circuits between Egypt and Europe, if that helps... Regards, Sahil Gupta VoiceValley On Thu, 15 Sep 2005, [ISO-8859-1] Stéphane LAVRI wrote: Hi I'm looking for a company who can provide me an Internet connection between africa and Europe. Plesa If someone can give me some contact name or company dont hesitate to send me a mail at [EMAIL PROTECTED] Best regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] internet connection between Africa and Europe
Hi I'm looking for a company who can provide me an Internet connection between africa and Europe. Plesa If someone can give me some contact name or company dont hesitate to send me a mail at [EMAIL PROTECTED] Best regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
>Could you point to a specific issue? Any chance it could be backported >to 1.0? This will mean less griff. http://www.google.ca/search?q=tdm+static+site:lists.digium.com&hl=en&lr=&rls =GGLD,GGLD:2004-23,GGLD:en&start=10&sa=N >When do you reboot? every day? Yes I do but the problem, for me, manifests itself every 1+N days so reboot every day preempts this behavior since it *will* run fine for at least a day. It's when I let it sit for 3 or 4 days that it happens. Zaptel reload works too, but might as well reboot since it's just sitting there from 10 PM to 6 AM. AFAIC, it's actually minor, we run 4 cordless phones that are used casually, a few times a day so no biggie if the problem happens during business hours. It does not affect any other operation of the server, SIP & IAX & SpanDSP, Sendmail, cdr, everything is 100%. This is the only fly in the ointment. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Oh323 and Asterisk with MERA
Sahil Gupta wrote: Client (MERA) --> H323 --> Asterisk --> IAX --> Asterisk You don't specify which H.323 channel driver you are using; there are least four possibilities at this time, so that would be helpful information. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Can not get realtime static voicemail.conf to work
Here is what happens on startup: Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime: Static SQL: SELECT category, var_name, var_val, cat_metric FROM settings WHERE filename='voicemail.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime: Everything is fine. Sep 15 13:23:51 DEBUG[28130] res_config_mysql.c: MySQL RealTime: Found 13 rows. Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: VM Review Option disabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: VM Operator break disabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: VM CID Info before msg disabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: Send Voicemail msg disabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: ENVELOPE before msg enabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: Duration info before msg enabled globally Sep 15 13:23:51 DEBUG[28130] app_voicemail.c: We are not going to skip to the next msg after save/delete But that's not what I have defined in my voicemail database: mysql->asterisk_vm->settings ++++---++--+--+-+ | id | cat_metric | var_metric | commented | filename | category | var_name | var_val | ++++---++--+--+-+ | 1 | 0 | 0 | 0 | voicemail.conf | default | format | wav|gsm | | 2 | 0 | 0 | 0 | voicemail.conf | default | attach | yes | | 3 | 0 | 0 | 0 | voicemail.conf | default | pbxskip | yes | | 4 | 0 | 0 | 0 | voicemail.conf | default | serveremail | [EMAIL PROTECTED] | | 5 | 0 | 0 | 0 | voicemail.conf | default | fromstring | xx Digital Voicemail | | 6 | 0 | 0 | 0 | voicemail.conf | default | maxmessage | 180 | | 7 | 0 | 0 | 0 | voicemail.conf | default | minmessage | 5 | | 8 | 0 | 0 | 0 | voicemail.conf | default | maxsilence | 5 | | 9 | 0 | 0 | 0 | voicemail.conf | default | silencethreshold | 128 | | 10 | 0 | 0 | 0 | voicemail.conf | default | skipms | 3000| | 11 | 0 | 0 | 0 | voicemail.conf | default | review | yes | | 12 | 0 | 0 | 0 | voicemail.conf | default | operator | no | | 13 | 0 | 0 | 0 | voicemail.conf | default | emailbody| You have a new voicemail! | ++++---++--+--+-+ Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Don't install asterisk-chan-capi
On Thu, Sep 15, 2005 at 04:48:04PM +0200, [EMAIL PROTECTED] wrote: > Show this message > adduser: Warning: The home dir you specified already exists > adduser: The user 'asterisk' already exist, and is non a system user. > dpkg: errrore processando asterisk-cah-capi (--configure): > il sottoprocesso post-installation script ha restituito un codice di errore 1 Do you refer to the Debian package? If so, could you please file a bug report (using reportbug) and/or provide more detials about the installed packages? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
On Thu, Sep 15, 2005 at 10:04:15AM -0600, Colin Anderson wrote: > Great comments everyone thanks and thanks for not flaming me. > > >Rebooting indicates that there is a problem that needs to be addressed. It > > >has nothing to do with uptime wars but with reliability and stability. I > >don't care if the system's down for 3 minutes due to reboot, I am concerned > > >that there is an underlying issue that you're merely masking by rebooting. > > In my case, I have the TDM04 static problem and that is the sole reason why > I reboot. > Granted, I'm running 1.0-stable and I understand it's fixed in -head or 1.2 > or whatever > but I don't want to go through the grief of upgrading to fix it and find out > that other issues are introduced. Could you point to a specific issue? Any chance it could be backported to 1.0? This will mean less griff. > A lot of guys say "upgrade, upgrade it's > way better" but then I read posts like the gentleman this morning who has > his console going crazy when he upgraded. My plan is to wait out the rest > of the year until 1.2 has a few months under it's belt then upgrade (hey, > isn't today the release day?). > Until then, reboot to me is a perfectly acceptable alternative and it does > not impact business operations one whit. You may feel differently, and yes, > I concede that I am masking an underlying problem, > but I am comfortable with it, my Asterisk server does not care, and it's > transparent to my users,so why not? When do you reboot? every day? This is assuming that the problem hits you randomly, you will rebooot some time after the problem has begun. e.g: there is a resonable chance that around half of the time the problem will exist. The more frequent the reboots are the less chances (but not 0%) are for "bad voice", but more time your PBX is unavailble. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Polycom oddities: Mixed up digits -> *8 Call Pickup
Hi, Last night I could dial *8 and pickup a call that was ringing to another phone. This morning, I searched on the Web for a solution to mixed up digits when dialing on a Polycom Soundpoint 501. I found that if you go to the SIP page on the phone's Web interface and change the "Digitmap Impossible Match" setting from "0" to "2" that fixes the mixed up/eaten digits problem. However, now *8 will not pick up a call. Has anyone had experience with these problems? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Fax->Email for Hosted PBX
I would agree wholeheartedly with everything Colin just said. I've had extensive experience with SpanDSP and with routing thru to an ATA. Both are touchy and work OK, but not well enough to make the users stop calling me :) Currently, we use SpanDSP and it works OK - although sometimes pages get mangled. I would *highly* recommend just getting a couple traditional POTS lines for faxes - it's still the most reliable way to go and what users are familiar with. You could hook a hylafax box up to those lines and still use a fax->email gateway. -- Tom Hayden On 9/15/05, Colin Anderson <[EMAIL PROTECTED]> wrote: > We use SpanDSP to recieve faxes on our PRI, a couple hundred a day with a > failure rate of ~5% which is pretty good I think but enough to tick people > off. Always the same fax numbers fail. What I did is have an exception list > context that is run just before RxFax. If the caller ID matches a "bad" > machine, the call is rerouted to a regular fax not off of a TDM or ATA but > forwarded to a POTS line (actually another channel on the same PRI - love > that Adtran!) > > In my tests, routing thru VoIP or out thru a TDM or ATA just plain doesn't > work, or doesn't work enough for it to be usable. You wold serve your > customer best by having Asterisk with SpanDSP at the telco demarc rx the fax > directly, and avoid shunting the call to an ATA or TDM. > > > -Original Message- > From: Alexander Lopez [mailto:[EMAIL PROTECTED] > Sent: Thursday, September 15, 2005 9:43 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Fax->Email for Hosted PBX > > > Best scenario does not route faxes over the IP network as a VoIP call. > > You can either use spandsp as a fax on the Asterisk box, (has problems, > but the delveloper is behind solving them) > > You can route the calls to a fax server located in the same colo via > tdm. (you can use HylaFax on Linix of any other solution. > > I have used Fax over VoIP for about 2-3 years. Some machines can't > handle it. Others can. Receiving has a higher success rate than > sending > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Michael > Welter > Sent: Thursday, September 15, 2005 11:09 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Fax->Email for Hosted PBX > > I'm proposing to install an Asterisk PBX at a collocation facility for a > > remote customer. Each of the customer locations will have an SPA-3000 > with the FXO port connecting a POTS circuit and the FXS port connecting > a fax machine or red phone. > > In addition to voice traffic, the customer has a high volume of incoming > > and outgoing faxes. > > Would it be possible, using g711 between the SPA-3000 and server, to > have spandsp/rxfax receive a fax from the POTS circuit via the SPA-3000? > > From the locally attached fax machine? (I realize that packet loss > will have a adverse effect on fax transmissions.) > > Would I be better-off attaching the fax machines to a Mediatrix 2102? > > Any help is appreciated. > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk CDR information into Oracle DB
On 17:04, Thu 15 Sep 05, Han van Hulst wrote: > What is the best way to get my CDR information into Oracle? > Is yada there for the best choice? > I would use ODBC. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?" ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] MusicOnHold not working
I have the same "problem" with several softphones (Xlite), but there's
one, Firefly I think, that worked. I found it strange, but not a real
problem for me. I have the same asterisk server version, wich came with
the last [EMAIL PROTECTED] distribution.On 9/15/05, Sherwood McGowan <[EMAIL PROTECTED]> wrote:
It's because mpg123 is being passed an option --mono. Looks to me (a cursoryguess) that your asterisk installation is trying to force mono sound, and
mpg123 doesn't like it.->-Original Message-->From: [EMAIL PROTECTED]->[mailto:
[EMAIL PROTECTED]] On Behalf Of->Gurminder Arora->Sent: Thursday, September 15, 2005 8:30 AM->To: asterisk-users@lists.digium.com
->Subject: [Asterisk-Users] MusicOnHold not working->->Hi-> On my FC3 box with asterisk 1.0.9MusicOnHold is not working.->It starts and stops immediately...->An unknow option mono comes...from where it is originating.??
->As there is nothing written in .conf file.->Console output is below:->->->I am using mpg123 version 0.59r.->Although I am able to play music with mpg123 but why it is on->No-cooperation movement against asterisk ?
->->Need help..any ideas any direction...->->Thanks->Gurminder->->->Console output->***Snip***->-- Executing MusicOnHold("Zap/1-1", "default") in new stack
->-- Started music on hold, class 'default', on channel 'Zap/1-1'->-- Stopped music on hold on Zap/1-1->Unknown option: --mono->***
->->->->My musiconhold.conf is->***Snip->[default]->mode=mp3->directory=/var/lib/asterisk/mohmp3->application=/usr/local/bin/mpg123
->Snip**->___->--Bandwidth and Colocation sponsored by Easynews.com --->
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RE: [Asterisk-Users] Fax->Email for Hosted PBX
We use SpanDSP to recieve faxes on our PRI, a couple hundred a day with a failure rate of ~5% which is pretty good I think but enough to tick people off. Always the same fax numbers fail. What I did is have an exception list context that is run just before RxFax. If the caller ID matches a "bad" machine, the call is rerouted to a regular fax not off of a TDM or ATA but forwarded to a POTS line (actually another channel on the same PRI - love that Adtran!) In my tests, routing thru VoIP or out thru a TDM or ATA just plain doesn't work, or doesn't work enough for it to be usable. You wold serve your customer best by having Asterisk with SpanDSP at the telco demarc rx the fax directly, and avoid shunting the call to an ATA or TDM. -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Thursday, September 15, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Fax->Email for Hosted PBX Best scenario does not route faxes over the IP network as a VoIP call. You can either use spandsp as a fax on the Asterisk box, (has problems, but the delveloper is behind solving them) You can route the calls to a fax server located in the same colo via tdm. (you can use HylaFax on Linix of any other solution. I have used Fax over VoIP for about 2-3 years. Some machines can't handle it. Others can. Receiving has a higher success rate than sending -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Thursday, September 15, 2005 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Fax->Email for Hosted PBX I'm proposing to install an Asterisk PBX at a collocation facility for a remote customer. Each of the customer locations will have an SPA-3000 with the FXO port connecting a POTS circuit and the FXS port connecting a fax machine or red phone. In addition to voice traffic, the customer has a high volume of incoming and outgoing faxes. Would it be possible, using g711 between the SPA-3000 and server, to have spandsp/rxfax receive a fax from the POTS circuit via the SPA-3000? From the locally attached fax machine? (I realize that packet loss will have a adverse effect on fax transmissions.) Would I be better-off attaching the fax machines to a Mediatrix 2102? Any help is appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] RxFax/TxFax - Compile Problem
I used the latest version (.3) and also the previous .2 ver (pre20). The spandsp seems to compile but when I download the rxfax/txfax .c files and drop them in the apps directory that is where I get the compile error. Dave From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, September 14, 2005 2:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RxFax/TxFax - Compile Problem What version of spandsp are you attempting to compile in to the 1.0.9 tree? On 9/14/05, David Sampson <[EMAIL PROTECTED]> wrote: Anyone know how to fix this? gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported only once /usr/include/asterisk/lock.h:302: error: for each function it appears in.) app_rxfax.c: In function `rxfax_exec': app_rxfax.c:263: warning: passing arg 1 of `fax_init' from incompatible pointer type app_rxfax.c:264: error: structure has no member named `verbose' app_rxfax.c:325: warning: passing arg 1 of `fax_release' from incompatible pointer type make[1]: *** [app_rxfax.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
On Thursday 15 September 2005 11:38, Paul wrote: > They designed it to be shut down. I guess that means it doesn't just > roll over like a dead cow. Actually dead cows aren't back-heavy. They typically just keep whatever position they were in when they took their last breath, much like the telco crosspoint switches. :-) > You wouldn't design a crane controller so that it releases the load on > reboot and then tries to return the cable to the pre-reboot position. > The same principle applies here. If it's mission-critical, you might > have to spend more on the hardware. A good example would be crosspoint I agree -- You need to spend more time on both hardware and software. For me, a phone system is like a file server or network switch -- it is *not* meant to be rebooted. If it needs to be periodically power-cycled then address the problem, don't simply put a cron script in. > I did hardware and software design for some devices that were usually > placed in very remote areas(underwater, mountains, arctic are a few). We > used a very low power clock device that would boot the cpu up. We did > whatever needed to be done, set the clock registers for the next wakeup > and shut down again. There were other ways to approach the problem, but > this approach made coding easier, used less power and allowed us to get > more functionality without increasing rom or ram size. If you only How did it save you ROM/RAM? I can see it saving having to put fancy power controller code and hardware in... Is that what you meant? > needed to take one simple measurement and store it to the ram, you could > do a boot/run/shutoff every second and still achieve some additional > battery runtime. Agreed but again -- you designed for this specific purpose. Asterisk isn't meant to boot up, answer the phone, process the call and shut down again until the next ring. (This would be an interesting approach to power savings though if your system boot time was fast enough and call volumes varied enough to make it worthwhile.) -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] RE: Stupid tricks: preventable?
>7. RE: Stupid tricks: preventable? (Colin Anderson)
Colin Anderson <[EMAIL PROTECTED]> wrote:
> i think you need a restart, then:
>
> [your-local-extension-context]
>
> exten => _,1,Gotoif(["${CALLERIDNUM}"="${EXTEN}"]?2:4)
> exten => _,2,Playback(you-are-a-frigging-idiot-stop-that)
> exten => _,3,System(/etc/asterisk/email-administrator-moronic-behavior
> ${CALLERIDNUM})
> exten => _,4,InsertNormalDialingBehaviorHere
Heh, good solution.
> Was this guy just trying to be a smart-alek?
Something like that, I think.
> I can't think of a reason why someone would want to dial their own
> extension from their own extension, let alone transfer it, unless they
> want to leave themselves voicemail??
That's good, the first reason I've heard of to call yourself at all.
In this case, on priority 2 above, I'd probably just send them to
Voicemail.
Though the whole "agent login" issue complicates the test
somewhat as well.
Thanks,
Alan
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[Asterisk-Users] Seperate Incoming calls on TDM02?
I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file to do this as it is my understanding that the control can be taken care of there. My system is running [EMAIL PROTECTED] 1.5. Help! Hatton ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
Great comments everyone thanks and thanks for not flaming me. >Rebooting indicates that there is a problem that needs to be addressed. It >has nothing to do with uptime wars but with reliability and stability. I >don't care if the system's down for 3 minutes due to reboot, I am concerned >that there is an underlying issue that you're merely masking by rebooting. In my case, I have the TDM04 static problem and that is the sole reason why I reboot. Granted, I'm running 1.0-stable and I understand it's fixed in -head or 1.2 or whatever but I don't want to go through the grief of upgrading to fix it and find out that other issues are introduced. A lot of guys say "upgrade, upgrade it's way better" but then I read posts like the gentleman this morning who has his console going crazy when he upgraded. My plan is to wait out the rest of the year until 1.2 has a few months under it's belt then upgrade (hey, isn't today the release day?). Until then, reboot to me is a perfectly acceptable alternative and it does not impact business operations one whit. You may feel differently, and yes, I concede that I am masking an underlying problem, but I am comfortable with it, my Asterisk server does not care, and it's transparent to my users,so why not? >Would you accept the need for a weekly reboot of your Nortel Option 11? How >about if your car required you to remove the battery for two minutes once a >month? Your VCR? How about a clock radio that had to be unplugged once a >week to fix weird little issues? See above. Yes I would accept it, provided that it would not impact normal operation and I as the administrator *or* enduser would not have to do anything special. However, *this* makes my sphincter shrink: http://slashdot.org/articles/02/07/22/0615221.shtml?tid=126 Obviously, that's no good. Best comment: RH support: Thanks for calling Red Hat! How may we help you? Pilot: "Uhh.. I'm spiraling towards the earth, both my engines are out, and my display says 'kernel panic' in white text on a black background." RH Support: "And what is the system model?" Pilot: "The F-22 jet.." RH support: If you read linux-kernel-bugtraq, you will see that you should have patched your kernel to 2.4.19-pre-alpha-revision-d before takeoff. But no problem, this is Linux after all. Do you have another F22 on your LAN? Just telnet in from there, su to root and restart sendmail. Pilot: @#$*! Redhat! I'm switching to Debian if I survive! >Hmm, I guess I won't be buying any Mitel equipment. Please don't. They are evil. Not so much the equipment but the corporate philosophy is something along the lines of "We're gonna hose our endusers more than any other telecom manufacturer" ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Comfort Noise Generation with Zap-IAX
Hello, we have a small Asterisk Network where Siemens PBX's are connected via PRI (Zap) to an Asterisk and the Asterisk's are connected through IAX, so this looks like this: Phone1 --- Siemens PBX --- Asterisk --- (IAX) --- Asterisk --- Siemens PBX --- Phone2 Now, when Phone1 calls Phone2 all wents well until there is silence - then the line seems to be death. My users wanted to have some Comfort Noise. The question is, how do I generate this comfort noise, and at which point? On Goolge I found some hints, that comfort noise generation is a phone feature (regarding VoIP phones). But our phones aren't SIP or other VoIP phones but normal Siemens Optiset phones, and they can't generate comfort noise. I suspect that I must generate the comfort noise in the VoIP part of the connection, more precisely, in iax.conf. Any suggestions or hints? Regards, Henry ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Fax->Email for Hosted PBX
Best scenario does not route faxes over the IP network as a VoIP call. You can either use spandsp as a fax on the Asterisk box, (has problems, but the delveloper is behind solving them) You can route the calls to a fax server located in the same colo via tdm. (you can use HylaFax on Linix of any other solution. I have used Fax over VoIP for about 2-3 years. Some machines can't handle it. Others can. Receiving has a higher success rate than sending -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Thursday, September 15, 2005 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Fax->Email for Hosted PBX I'm proposing to install an Asterisk PBX at a collocation facility for a remote customer. Each of the customer locations will have an SPA-3000 with the FXO port connecting a POTS circuit and the FXS port connecting a fax machine or red phone. In addition to voice traffic, the customer has a high volume of incoming and outgoing faxes. Would it be possible, using g711 between the SPA-3000 and server, to have spandsp/rxfax receive a fax from the POTS circuit via the SPA-3000? From the locally attached fax machine? (I realize that packet loss will have a adverse effect on fax transmissions.) Would I be better-off attaching the fax machines to a Mediatrix 2102? Any help is appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Getting email of voicemail to work
[EMAIL PROTECTED] uses Sendmail. The default configuration of AAH's sendmail is not to relay the email to an external server, but to try local delivery. You must modify the sendmail.mc file in /etc/mail with this line: define(`SMART_HOST',`mail-out.your.provider') where `mail-out.your.provider' is the fully qualified name of your regular SMTP server. You must also configure your SMTP server to allow relay connections from the IP address of your Asterisk server. This varies, depending on what kind of SMTP server you have. Once you are done editing sendmail.mc, run "make" from /etc/mail and run "service sendmail restart" To test it, "sendmail -v [EMAIL PROTECTED]" from the command line. Type some crap, press enter, type a period, press enter. You will see some stuff printed to the console finally ending in something like: "Message accepted for delivery". You should see an email message from "root" (or whatever the curently logged on user is) in your mailbox This is the 2 cent version, Sendmail is a frustrating and bitchy beast to get working properly but once it's working, nothing beats it. Google "Sendmail smart host howto" or see this link: http://www.tldp.org/HOWTO/Sendmail-Address-Rewrite-3.html hth -Original Message-From: Matt Love [mailto:[EMAIL PROTECTED]Sent: Thursday, September 15, 2005 9:19 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Getting email of voicemail to work Hi, Can someone point me in the direction of getting the voicemail -> Email to work on [EMAIL PROTECTED] 1.5 Ive put in the email addresses of voicemail users eg [EMAIL PROTECTED] But i cant find where to set the email server up. we have a company email server an idealy i would like to relay the mails to it. However i cant find within the AMP where to setup either the Sendmail server or put in a relay address for an external server. Can anyone help. (you may have guessed im an @ newbie) Thanks Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
Andrew Kohlsmith wrote: Hmm, I guess I won't be buying any Mitel equipment. The MARS rovers were designed to be totally shut down as a last measure to ensure everything is starting up as they'd simulated on Earth and that there was no high-energy radiation glitches due to space travel. They designed it to be shut down. I guess that means it doesn't just roll over like a dead cow. You wouldn't design a crane controller so that it releases the load on reboot and then tries to return the cable to the pre-reboot position. The same principle applies here. If it's mission-critical, you might have to spend more on the hardware. A good example would be crosspoint switch modules that don't change state when the system is rebooted. That would prevent calls from being dropped as long as you don't design your system software to explicity clear all those modules at startup. I did hardware and software design for some devices that were usually placed in very remote areas(underwater, mountains, arctic are a few). We used a very low power clock device that would boot the cpu up. We did whatever needed to be done, set the clock registers for the next wakeup and shut down again. There were other ways to approach the problem, but this approach made coding easier, used less power and allowed us to get more functionality without increasing rom or ram size. If you only needed to take one simple measurement and store it to the ram, you could do a boot/run/shutoff every second and still achieve some additional battery runtime. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Transfering from a device to a queue crashes Asterisk
Asterisk crashes with no errors when I transfer from a device (my phone) to a queue Asterisk crashes with no errors. Also if I xfer from a sip device to another and dont wait for the other user to pickup before xfering the call gets dropped. Any ideas? Im using the latest cvs of asterisk, amp 1.10.009 and our phones are polycom 501. Yahoo! for Good Click here to donate to the Hurricane Katrina relief effort. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Getting email of voicemail to work
Hi, Can someone point me in the direction of getting the voicemail -> Email to work on [EMAIL PROTECTED] 1.5 Ive put in the email addresses of voicemail users eg [EMAIL PROTECTED] But i cant find where to set the email server up. we have a company email server an idealy i would like to relay the mails to it. However i cant find within the AMP where to setup either the Sendmail server or put in a relay address for an external server. Can anyone help. (you may have guessed im an @ newbie) Thanks Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo on SPA-3000 FXO
Title: Message
The
SPA-3000 can do silence and tone detection for hangup and has a variable timer
and sound threshold setting.
-Original Message-From: BJ Weschke
[mailto:[EMAIL PROTECTED] Sent: Thursday, September 15, 2005 10:01
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Echo on SPA-3000
FXO That is an issue with Vonage not providing remote
disconnect supervision through the ATA. The way I got around that was not
to use Comedian mail with my home system, but instead, just use an analog
answering machine I already had around. Not ideal, I know, but the answering
machine hangs up the FXS port when it detects that the caller has hung up (via
busy, silence, etc) and that instructs * to hang up the FXO port that has the
Vonage ATA plugged into it.
On 9/15/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I
tried switching out the 3000 with a X100P card, but the card would never
recognize when the caller hung up. So, I have to keep the 3000.
The X100P worked fine with the POTS line.
At 08:52 AM 9/15/2005, you
wrote:
The CID with the Cisco isn't a
"Cisco issue". It's actually an issue based on the way Vonage passes CID
through the Cisco. It doesn't follow the same standard that LECs and others
use. I tried to get this going with an SPA3000 at first
as well and never really could get it to go right without having to really
hack-up my chan_sip.c. In the end I switched out the SPA-3000 for a Digium
TDM card with an FXO module on it and now use the following in the dial plan
for when that fxo port rings: [fxoports]exten =>
s,1,Cut(cidone=CALLERID,<,1)exten =>
s,2,Cut(cidtwo=CALLERID,<,2)exten =>
s,3,GotoIf($[${LEN(${cidtwo})} > 0]?4:12)exten =>
s,4,SetCIDNum(${cidtwo:1:10})exten => s,5,SetVar(cidone=${cidone:2})
exten => s,6,NoOp(${cidone})exten =>
s,7,SetVar(ciolen=${LEN(${cidone})})exten =>
s,8,SetVar(ciolen=$[${ciolen} - 2])exten =>
s,9,NoOp(${ciolen})exten => s,10,SetCIDName(${cidone:0:${ciolen}})
exten => s,11,Goto(fxoports,s,14)exten =>
s,12,SetCIDNum(${CALLERID})exten =>
s,13,SetCIDName(Unavailable)exten =>
s,14,NoOp(${CALLERIDNUM})exten =>
s,15,NoOp(${CALLERIDNAME}) CVS-HEAD versions of asterisk will
complain when you do this because it's using the deprecated SetCIDNum and
SetCIDName functions. In that case you'll want to replace those with SetVar(CALLERID(name)=value) and SetVar(CALLERID(number)=value)
instead. On 9/15/05, [EMAIL PROTECTED]
<[EMAIL PROTECTED]>
wrote:
I've been using 2 SPA3000's for several months. Both are
running
3.1.3(GWa) software. I do not have any issues with echo.
One box is
used to bring in a SBC POTS line and the other is connected to my
Cisco ATA186 from Vonage. The 3000 connected to SBC line relays
CID
info, I have never been able to get CID info passed from the 3000
connected to the Cisco unit.
Doug
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[Asterisk-Users] Siemens Hi-Path help
hi all, anybody have a siemens hipath 3500 with a sm2/pri card? because i need to connect to my box TE110P (e1) and i dont know how is the mode in the pbx to change it. thanks -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Fax->Email for Hosted PBX
I'm proposing to install an Asterisk PBX at a collocation facility for a remote customer. Each of the customer locations will have an SPA-3000 with the FXO port connecting a POTS circuit and the FXS port connecting a fax machine or red phone. In addition to voice traffic, the customer has a high volume of incoming and outgoing faxes. Would it be possible, using g711 between the SPA-3000 and server, to have spandsp/rxfax receive a fax from the POTS circuit via the SPA-3000? From the locally attached fax machine? (I realize that packet loss will have a adverse effect on fax transmissions.) Would I be better-off attaching the fax machines to a Mediatrix 2102? Any help is appreciated. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk CDR information into Oracle DB
What is the best way to get my CDR information into Oracle? Is yada there for the best choice? How stable is yada for this Thanks Han ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo on SPA-3000 FXO
That is an issue with Vonage not providing remote disconnect supervision through the ATA. The way I got around that was not to use Comedian mail with my home system, but instead, just use an analog answering machine I already had around. Not ideal, I know, but the answering machine hangs up the FXS port when it detects that the caller has hung up (via busy, silence, etc) and that instructs * to hang up the FXO port that has the Vonage ATA plugged into it.
On 9/15/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I tried switching out the 3000 with a X100P card, but the card would never recognize when the caller hung up. So, I have to keep the 3000. The X100P worked fine with the POTS line.
At 08:52 AM 9/15/2005, you wrote:
The CID with the Cisco isn't a "Cisco issue". It's actually an issue based on the way Vonage passes CID through the Cisco. It doesn't follow the same standard that LECs and others use.
I tried to get this going with an SPA3000 at first as well and never really could get it to go right without having to really hack-up my chan_sip.c. In the end I switched out the SPA-3000 for a Digium TDM card with an FXO module on it and now use the following in the dial plan for when that fxo port rings:
[fxoports]exten => s,1,Cut(cidone=CALLERID,<,1)exten => s,2,Cut(cidtwo=CALLERID,<,2)exten => s,3,GotoIf($[${LEN(${cidtwo})} > 0]?4:12)exten => s,4,SetCIDNum(${cidtwo:1:10})
exten => s,5,SetVar(cidone=${cidone:2}) exten => s,6,NoOp(${cidone})exten => s,7,SetVar(ciolen=${LEN(${cidone})})exten => s,8,SetVar(ciolen=$[${ciolen} - 2])exten => s,9,NoOp(${ciolen})exten => s,10,SetCIDName(${cidone:0:${ciolen}})
exten => s,11,Goto(fxoports,s,14)exten => s,12,SetCIDNum(${CALLERID})exten => s,13,SetCIDName(Unavailable)exten => s,14,NoOp(${CALLERIDNUM})exten => s,15,NoOp(${CALLERIDNAME}) CVS-HEAD versions of asterisk will complain when you do this because it's using the deprecated SetCIDNum and SetCIDName functions. In that case you'll want to replace those with
SetVar(CALLERID(name)=value) and SetVar(CALLERID(number)=value) instead. On 9/15/05,
[EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I've been using 2 SPA3000's for several months. Both are running
3.1.3(GWa) software. I do not have any issues with echo. One box is
used to bring in a SBC POTS line and the other is connected to my
Cisco ATA186 from Vonage. The 3000 connected to SBC line relays CID
info, I have never been able to get CID info passed from the 3000
connected to the Cisco unit.
Doug ___--Bandwidth and Colocation sponsored by
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[Asterisk-Users] Configuring GR303 trunks from Asterisk to a Taqua/TEKELEC T7000
We are trying to configure two GR303 trunks from an Asterisk server with a quad span card to a class 5 softswitch (Taqua OCX/TEKELEC T7000). We show the T1s up but errors on the TMC & EOC channels. Has anyone configured GR303 before and/or setup this type of configuration. Thanks! Paul ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Don't install asterisk-chan-capi
Show this message adduser: Warning: The home dir you specified already exists adduser: The user 'asterisk' already exist, and is non a system user. dpkg: errrore processando asterisk-cah-capi (--configure): il sottoprocesso post-installation script ha restituito un codice di errore 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk 1.0.9 long term stability <--thread hijack, why not reboot?
On Wednesday 14 September 2005 12:34, Colin Anderson wrote: > I'm curious as to this obsession with uptime is. All of the posts of this > type are along the lines of "After X days, Y thing does not work but if I > reload or reboot, it's OK" - so why not cron a reboot? Is it considered bad > form or something like that? I reboot every night whether it is needed or > not, not afraid to admit it, and everything works fine for me. Rebooting indicates that there is a problem that needs to be addressed. It has nothing to do with uptime wars but with reliability and stability. I don't care if the system's down for 3 minutes due to reboot, I am concerned that there is an underlying issue that you're merely masking by rebooting. > We also do the "Sunday reboot" of all of our Windows servers as well as > restarting all of the critical services such as IIS , SQL, Exchange etc > nightly. It helps, a lot (Exchange is a notorious memory leaker) Exactly. With closed-source proprietary applications you can't fix the problem so this is all you can do. With Asterisk and OSS you can fix it and increase code quality. Would you accept the need for a weekly reboot of your Nortel Option 11? How about if your car required you to remove the battery for two minutes once a month? Your VCR? How about a clock radio that had to be unplugged once a week to fix weird little issues? I put Asterisk into the same class as these kinds of devices. They must be up and stay up. If they need to be rebooted then there's an issue that needs to be addressed and corrected. > I've also heard it said, something along the lines of: "If you have to > reboot, your server isn't set up correctly" to which I say piffle. Even > NASA has rebooted the Mars probes after they land and I understand that > they run VXWorks, incidentally, the same RTOS that my Mitel 3300 uses, and > *even Mitel* recommends periodic reboots, which we duly cron every night, 2 > AM. Hmm, I guess I won't be buying any Mitel equipment. The MARS rovers were designed to be totally shut down as a last measure to ensure everything is starting up as they'd simulated on Earth and that there was no high-energy radiation glitches due to space travel. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] USB ISDN (OT question)
Derek, could you give me some details regarding the solar power supply you're using for your installation? Thanks! Jörg > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Derek Conniffe > Sent: Thursday, September 15, 2005 12:28 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] USB ISDN > > In a remote location (solar powered) I have an old Gateway > 333mhz notebook running Asterisk and using a PCMCIA AVM Fritz > ISDN card. With the AVM CAPI it works perfectly with chan_capi. > > Derek > > Julien Goodwin wrote: >> (...) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk CDR information into Oracle DB
What is the best way to get my CDR information into Oracle? Is yada there for the best choice? How stable is yada for this Thanks Han ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk don't start
it looks to me like you haven't loaded the zaptel.o or ztdummy.o kernel modules... On Thursday 15 September 2005 16:10, [EMAIL PROTECTED] wrote: > Asterisk don't running, because show this message > > WARNING[6949]: chan_sip.c:8865 reload_config: Section 'authentication' > lacks type > > WARNING[6949]: chan_iax2.c:7491 load_module: Unable to open IAX timing > interface: No such file or directory > > WARNING[6949]: chan_skinny.c:2587 reload_config: Unable to get our IP > address, Skinny disabled > > WARNING[6949]: chan_oss.c:239 sound_thread: Read error on sound device: > Resource temporarily unavailable > > WARNING[6949]: loader.c:258 ast_load_resource: > /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: > pri_suspend_acknowledge Sep 15 16:05:46 WARNING[6949]: loader.c:440 > load_modules: Loading module chan_zap.so failed! > > > Thanks! > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Grandstream HandyTone 386
Hello all I have a question about Grandstream HandyTone 386 can Grandstream HandyTone 386 make 2 sim. calls with g729 codec in same time Iyi Calismalar. Ugur GUNCER smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk don't start
Asterisk don't running, because show this message WARNING[6949]: chan_sip.c:8865 reload_config: Section 'authentication' lacks type WARNING[6949]: chan_iax2.c:7491 load_module: Unable to open IAX timing interface: No such file or directory WARNING[6949]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled WARNING[6949]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable WARNING[6949]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_suspend_acknowledge Sep 15 16:05:46 WARNING[6949]: loader.c:440 load_modules: Loading module chan_zap.so failed! Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] dialplan to try VOIP providers if they can't terminate call
I am trying to figure out how to try different VOIP providers if they aren't able to terminate the call because they don't offer service to that dialing area. The error that gets logged to the console is: Sep 13 00:01:43 WARNING[22093]: chan_iax2.c:6835 socket_read: Call rejected by x.x.x.x: No such context/extension The dialstatus returned is "No Answer" I could check for that dialstatus, but it is the same dial status as if someone didn't pick up the phone. How are ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Echo on SPA-3000 FXO
I tried switching out the 3000 with a X100P card, but the card would
never recognize when the caller hung up. So, I have to keep the
3000. The X100P worked fine with the POTS line.
At 08:52 AM 9/15/2005, you wrote:
The CID with the Cisco
isn't a "Cisco issue". It's actually an issue based on the way
Vonage passes CID through the Cisco. It doesn't follow the same standard
that LECs and others use.
I tried to get this going with an SPA3000 at first as well and
never really could get it to go right without having to really hack-up my
chan_sip.c. In the end I switched out the SPA-3000 for a Digium TDM card
with an FXO module on it and now use the following in the dial plan for
when that fxo port rings:
[fxoports]
exten => s,1,Cut(cidone=CALLERID,<,1)
exten => s,2,Cut(cidtwo=CALLERID,<,2)
exten => s,3,GotoIf($[${LEN(${cidtwo})} > 0]?4:12)
exten => s,4,SetCIDNum(${cidtwo:1:10})
exten => s,5,SetVar(cidone=${cidone:2})
exten => s,6,NoOp(${cidone})
exten => s,7,SetVar(ciolen=${LEN(${cidone})})
exten => s,8,SetVar(ciolen=$[${ciolen} - 2])
exten => s,9,NoOp(${ciolen})
exten => s,10,SetCIDName(${cidone:0:${ciolen}})
exten => s,11,Goto(fxoports,s,14)
exten => s,12,SetCIDNum(${CALLERID})
exten => s,13,SetCIDName(Unavailable)
exten => s,14,NoOp(${CALLERIDNUM})
exten => s,15,NoOp(${CALLERIDNAME})
CVS-HEAD versions of asterisk will complain when you do this
because it's using the deprecated SetCIDNum and SetCIDName functions. In
that case you'll want to replace those with
SetVar(CALLERID(name)=value) and
SetVar(CALLERID(number)=value) instead.
On 9/15/05,
[EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
I've been using 2 SPA3000's for several months. Both are
running
3.1.3(GWa) software. I do not have any issues with echo.
One box is
used to bring in a SBC POTS line and the other is connected to
my
Cisco ATA186 from Vonage. The 3000 connected to SBC line relays
CID
info, I have never been able to get CID info passed from the
3000
connected to the Cisco unit.
Doug
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Re: [Asterisk-Users] PSTN calls are quiet
I just did it by ear. Got it right in less than 5 minutes. --- Paul Goodyear <[EMAIL PROTECTED]> wrote: > I thought the txgain, rxgain was purely for echo settings. > > Is there a rough guide to this process, or is it a simple case of > changing values and testing them? > > Thanks. > > On 9/15/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > > Hi Paul > > > > There are two settings in zapata.conf called txgain and rxgain. You can > set > > these to adjust the volume on your PSTN lines. They can be set in db or > as a > > percentage. > > > > Garth > > > > > > > > > > --- Paul Goodyear <[EMAIL PROTECTED]> wrote: > > > Sip to sip calls are fine, both local on Asterisk and over a SIP > > > gateway, however some people who call on the PSTN line say we are very > > > queit and vice versa, can the volume be turned up on the PSTN line? > > > > > > The volume buttons on the VoIP phones only turns up the others voice, > > > so this is a fix for us, but how do I make our voices louder for the > > > people on the PSTN line? > > > > > > Thanks. > > > > > > Paul. > > > ___ > > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > > > 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 > > > > > > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > 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 > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] cdr server
I send CDR to a MySQL database on another machine. Might that work for you? --On Thursday, September 15, 2005 3:24 PM +0200 Altus Snyman <[EMAIL PROTECTED]> wrote: Good day all Is it possable to set asterisk up as a cdr server for other voip units We got a quintum dx here and its got a option to log to a cdr server on port 9002 Thanks Altus ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
