Re: [Asterisk-Users] Snom 360 Opinions
I'm not too pleased with the phones, I have about 40 of them, some of the displays tend to die and the dial pad feels to 'mushy' IMHO, just like the keys on a good old ZX80 computer Also I'm having some issues with sound quality on some phones, but I still need to switch some phones to see if that is really an issue of the phone. Also if you want to use * call files, with the 360 you will run into a big where the call is being redialled as if it failed while in fact the call is ongoing. Annoying and haven't found out if that is an * bug or Snom bug. The Snom 190's do not have this problem. Just my $0.02 (which is really not a lot these days!) :) On Sat, 12 Nov 2005, Curren C. Calhoun wrote: I¹m looking to add in some Snom 360 phones, could anyone give thoughts or opinions about the speakerphone, general quality... Also the phone would need to be powered over Ethernet... I like some of the listed features and the expandability of the phone but am open to any other suggestions as well... Thanks Curren ___ --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] callcentrum - call any, ring one
I would like, that incomming calls displays (clid) on all ip phones in group (helpdesk), but only on user specific extension ringing... e.g. helpdesk employee have number 310, 311, 312 etc., when incomming call to 311 line, only 311 line ringing, but on all other phones (310, 312) only display caller id, so that other emploee can eventualy pickup... maybe use of dial with ALERT_INFO variable specifiing silent_ring.wav to quietly call other phones? any better idea? e.g. special use of queue app? PJ ___ --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] How to let caller continue after Dial cmd
We have a need to allow the caller who is in the middle of a call (i.e. who is already bridged between a PRI channel and a SIP channel as the result of entering a Dial cmd in the current context) to type something like ## to cause the called party to be disconnected and to return from the Dial command with a distinctive STATUS so they can proceed to do other things within the context. Sort of a combination of the H and g flags to the Dial cmd. Any thoughts about whether or not this is possible in 1.0 or 1.2? If not, is there sufficient interest in such a feature for us to submit it once complete (I don't want to go through the effort to properly document, post, and maintain such a patch if its an improvement no one wants). -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --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] Intel Desktop MotherBoards Unsuitable for DigiumBoards
Kevin P. Fleming wrote: 'lspci -vb' does not understand IO-APIC mode, as best I can tell, so the interrupt number that it reports is totally useless. On my desktop machine, with an nVidia graphics card in a PCI-Express slot, /proc/interrupts shows it using interrupt 185, and 'lspci -vb' shows it using IRQ 10. It is most definitely _not_ on IRQ 10, since /proc/interrupts show the actual mapping from the IRQ controller (the APIC) to the driver servicing the device. At best, 'lspci -vb' is showing what interrupt the device was on _before_ the kernel reassigned it using APIC mode. Mr. Fleming: I am aware of all of the above but does the fact you are posting this mean that Digium is now aware of this? I have had a Digium tech ssh to a machine, run an lspci -vb, report your interrupts are shared and refuse to work on the case until the lspci -vb showed unique IRQs. Does your post mean that this policy has now changed at Digium? As Piotr A. Sygula wrote in another related post: I.e. although APIC is splitting up IRQ's rather nicely, the tech support guy is saying that it doesn't matter what the APIC layer says. Would someone out there break the tie? I'd like an educated opinion/statement on whether APIC support solves the IRQ sharing issue, or simply masks it. Which suggests other customers have been told by Digium that lspci -vb is the final word on whether IRQs are being shared and whether or not your hardware configuration is supported by Digium. So what's the real story -- does APIC count or not? Will Digium support systems with unique APIC interrupts but not unique lspci -vb interrupts? -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --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 :Re: [Asterisk-Users] Softphone with Lotus Notes support?
Hi, Do you think there would be any interest in a softphone that supports LDAP ? why not? You can use ldap commands to connect to Domino and MS ADS, so a softphone with ldap capabilities sounds like quite a good idea to me. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support ___ --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] Capi problem
Hi Guys Im having a problem getting CAPI to work on my Traverse NetJet card. CAPI is enabled in the kernel and Im using the mISDN drivers with the NetJet patch. I cant seem to get astcapi to load Heres the output im getting Nov 12 21:18:36 VERBOSE[4011]: == Registered application 'WaitMusicOnHold' Nov 12 21:18:36 VERBOSE[4011]: == Registered application 'SetMusicOnHold' Nov 12 21:18:36 VERBOSE[4011]: [chan_capi.so]Nov 12 21:18:36 VERBOSE[4011]: [chan_capi.so] = (Common ISDN API for Asterisk) Nov 12 21:18:36 VERBOSE[4011]: == Parsing '/etc/asterisk/capi.conf': Nov 12 21:18:36 VERBOSE[4011]: == Parsing '/etc/asterisk/capi.conf': Found Nov 12 21:18:36 VERBOSE[4011]: -- ast_capi_pvt ISDN1 (*,from-pstn,0,2) (0,4,64) Nov 12 21:18:36 VERBOSE[4011]: -- ast_capi_pvt ISDN1 (*,from-pstn,0,2) (0,4,64) Nov 12 21:18:36 WARNING[4011]: CAPI not installed, CAPI disabled! Nov 12 21:18:36 WARNING[4011]: chan_capi.so: load_module failed, returning -1 Nov 12 21:18:36 VERBOSE[4011]: == Unregistered channel type 'CAPI' Nov 12 21:18:36 WARNING[4011]: Loading module chan_capi.so failed! I think it could be a udev problem so I put a file called 10-capi.rules in my udev directory with the following SYSFS(dev)=68:0, NAME=capi20 SYSFS(dev)=191:[0-9]*, NAME=capi/%n When I do a capiinit it says ERROR: cannot load module kernelcapi Does there need to be another entry for kernelcapi? Any help would be greatly 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
Re: [Asterisk-Users] Problems after upgrade...
On Nov 12, 2005, at 12:11 AM, Francois Meehan wrote: Hi all, I have upgrade my kernel and asterisk to their latest release on a Centos 4.1 box, now it won't start anymore. Have you rebuilt Zaptel against your new kernel? If you upgrade the kernel, you need to rebuild zaptel. 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] Snom 360 Opinions
Cleaner looking, 12 line apperances, affordable sidecar, runs on linux, developing XML services, very programable, buttons are firm in a good way, simple layout for users installing 30 right now... On 11/12/05, Remco Barende [EMAIL PROTECTED] wrote: I'm not too pleased with the phones, I have about 40 of them, some of the displays tend to die and the dial pad feels to 'mushy' IMHO, just like the keys on a good old ZX80 computer Also I'm having some issues with sound quality on some phones, but I still need to switch some phones to see if that is really an issue of the phone. Also if you want to use * call files, with the 360 you will run into a big where the call is being redialled as if it failed while in fact the call is ongoing. Annoying and haven't found out if that is an * bug or Snom bug. The Snom 190's do not have this problem. Just my $0.02 (which is really not a lot these days!) :) On Sat, 12 Nov 2005, Curren C. Calhoun wrote: I¹m looking to add in some Snom 360 phones, could anyone give thoughts or opinions about the speakerphone, general quality... Also the phone would need to be powered over Ethernet... I like some of the listed features and the expandability of the phone but am open to any other suggestions as well... Thanks Curren ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! --- ___ --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] PRI testing using TE205 and loopback cable?
Using fc3 with cvs-head from Nov 1st and TE205P (dual T1 port) with a T1 cross-over cable between span 1 and 2. (udev properly defined.) In /etc/zaptel.conf I have: span=1,0,0,esf,b8zs,yellow bchan=1-23 dchan=24 span=2,0,0,esf,b8zs,yellow bchan=25-47 dchan=48 And in zapata.conf I have: [channels] language=en context=pri-in signalling=pri_net switchtype=national pridialplan=unknown channel=1-23 context=pri-out switchtype=national signalling=pri_cpe pridailplan=unknown group=7 channel=25-47 When starting asterisk with 'asterisk -cvvvddd', I see: Nov 12 08:46:07 WARNING[6938]: chan_zap.c:8875 pri_dchannel: PRI Error: We think we're the network, but they think they're the network, too. Nov 12 08:46:07 WARNING[6939]: chan_zap.c:8875 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. Why the warnings? (The warnings keep repeating forever) ___ --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] Intel Desktop MotherBoards Unsuitable for DigiumBoards
George Pajari wrote: I am aware of all of the above but does the fact you are posting this mean that Digium is now aware of this? It means I am :-) I have been trying to teach the tech support team about this, but I don't think it has quite sunk in yet. I believe that means it's time for another training session... I have had a Digium tech ssh to a machine, run an lspci -vb, report your interrupts are shared and refuse to work on the case until the lspci -vb showed unique IRQs. I understand. Does your post mean that this policy has now changed at Digium? It is in the process of changing, yes. ___ --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] Quantumvoice vs Broadvoice - Multiline
Teliax looks good - not comfortable with the soft limits but love the free setup! [EMAIL PROTECTED] wrote: Have you looked into teliax? 4 simultaneous calls on a bus plan is pretty good for less than $50/mo. And I cannot complain about the quality or the support. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Saul Diaz Sent: Friday, November 11, 2005 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline Julio Arruda wrote: I was testing Broadvoice few weeks before Hurricane Wilma here in FL. Since then, I had been since the landline (Bellsouth), and I had to 'remote callfwd' the BS # to my broadvoice #. So, from my impression, is ok for my needs (I got a weird no ringback problem that I kind of solved with a Background trick), and no surprises yet regarding the bill (my mother in law call Brazil a lot from my house, no, she is not aware of the 'unlimited' plan. So I may be in for a surprise in a couple of months). I've no tried several calls at the same time, you may want to ask them.. PS: I'm running Asterisk 1.0.9 Dane Reugger wrote: We are considering Quantumvoice as a provider - They are telling us they will give us 1 line number but we can have 5 concurrent incoming and outgoing line numbers. Charge is about $45 + extras - this seems considerable less expensive than the competition which seem to focus on. My second choice is BroadVoice $29.99 + $9.99 per additional line (in state only?) - more expensive, less features, and they don't seem loved by many ? Is anyone else using Quantum Voice? It was mentioned earlier that it requires an ATA connection and Asterisk support/compatibility is sketchy at best - I've contacted BV and they responded saying they need 24hrs to look into it? Seems like a popular topic but I'm looking for 2-3 lines - I only need one number but need to be able to make or receive several calls at a time? Any advice or recommendations appreciated - I want to port my number but I'm running out of time and must make a decision very soon. Thanks, Dane Reugger Crescent City Technologies New Orleans, LA 70112 ___ --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 Broadvoice only allows only the normal 3 way calling so is 2 channels for # about BV i got a lot of water under the bridge every works ok supper ok for times. then BV brokes without you make a single change in your asterisk server and stop working.. if u call support you are the guy with the problem.. yes BV support sucks, and it took me 9 phone calls, 12 emails, 3 chargeback and 2 call to my bank to remove myself from their billing all them well documented... so my advice nothing can be worts than BV. regards Saul ___ --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] [EMAIL PROTECTED] KDE or GNOME?
Do [EMAIL PROTECTED] have KDE or GNOME? How to start GUI? ___ --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] [EMAIL PROTECTED] KDE or GNOME?
No it does not. You should not run X with asterisk. If you want answers to questions about AAH you should ask those questions on that forum. http://sourceforge.net/forum/?group_id=123387On 11/12/05, Goran [EMAIL PROTECTED] wrote:Do [EMAIL PROTECTED] have KDE or GNOME?How to start GUI? ___--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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Phone: 978-203-3848 x205Fax: 518-631-2856 ___ --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] Does IAX2 Trunk Work between IAX and SIP
Hi: Does Trunk=yes in IAX save bandwidth as it should in case the other server (voip provider) has SIP only? Regards; Chawki __ 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] REaltime does not unregister sip peers on the fly
Hi, chick*CLI show version files chan_sip.c File Revision chan_sip.cRevision: 1.907 chick*CLI show version files pbx_realtime.c File Revision pbx_realtime.cRevision: 1.15 chick*CLI show version Asterisk CVS HEAD built by root @ chick on a i686 running Linux on 2005-11-09 14:28:18 UTC extconfig.conf sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies extentions = mysql,asterisk,extensions_table Registered sip friends work great through realtime. However, setting in sip.conf: rtcachefriends=yes rtupdate=yes rtautoclear=yes ignoreregexpire=no ( trying to have show peers available, but attempting to clear the cache) and deleting the secret the friend , this friend :) stays registered: Seeding..., Saved... and all. Weren't they supposed with realtime to get busted on the fly at the next registration without sip reload? Or just created (on the fly)? Anybody? benchev ___ --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] Snom 360 Opinions
Thanks that's the type of info I'm looking for... I've heard some early grumblings but wanted to see if anything else has come up... From: Remco Barende [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 12 Nov 2005 09:50:01 +0100 (CET) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Snom 360 Opinions I'm not too pleased with the phones, I have about 40 of them, some of the displays tend to die and the dial pad feels to 'mushy' IMHO, just like the keys on a good old ZX80 computer Also I'm having some issues with sound quality on some phones, but I still need to switch some phones to see if that is really an issue of the phone. Also if you want to use * call files, with the 360 you will run into a big where the call is being redialled as if it failed while in fact the call is ongoing. Annoying and haven't found out if that is an * bug or Snom bug. The Snom 190's do not have this problem. Just my $0.02 (which is really not a lot these days!) :) On Sat, 12 Nov 2005, Curren C. Calhoun wrote: I¹m looking to add in some Snom 360 phones, could anyone give thoughts or opinions about the speakerphone, general quality... Also the phone would need to be powered over Ethernet... I like some of the listed features and the expandability of the phone but am open to any other suggestions as well... Thanks Curren ___ --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] PRI testing using TE205 and loopback cable?
Answering self... this is a remote box and the guy that was suppose to install the T1 crossover cable installed a loopback plug instead. Sorry for the noise. Using fc3 with cvs-head from Nov 1st and TE205P (dual T1 port) with a T1 cross-over cable between span 1 and 2. (udev properly defined.) In /etc/zaptel.conf I have: span=1,0,0,esf,b8zs,yellow bchan=1-23 dchan=24 span=2,0,0,esf,b8zs,yellow bchan=25-47 dchan=48 And in zapata.conf I have: [channels] language=en context=pri-in signalling=pri_net switchtype=national pridialplan=unknown channel=1-23 context=pri-out switchtype=national signalling=pri_cpe pridailplan=unknown group=7 channel=25-47 When starting asterisk with 'asterisk -cvvvddd', I see: Nov 12 08:46:07 WARNING[6938]: chan_zap.c:8875 pri_dchannel: PRI Error: We think we're the network, but they think they're the network, too. Nov 12 08:46:07 WARNING[6939]: chan_zap.c:8875 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. Why the warnings? (The warnings keep repeating forever) ___ --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 ---End of Original Message- ___ --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] problems compiling spandsp-0.0.2pre21c under 1.2rc2
Has anybody sucessfuly compilied spandsp-0.0.2pre21c under 1.2rc2? I keep getting this: [Nov 12 10:14:16] [app_rxfax.so][Nov 12 10:14:16] WARNING[12188]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler [Nov 12 10:14:16] WARNING[12188]: loader.c:554 load_modules: Loading module app_rxfax.so failed! [EMAIL PROTECTED] src]# output: fwrite: Broken pipe output: fwrite: Broken pipe output: fwrite: Broken pipe Anybody had better luck? ___ --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 play dialtone
Hiya I've had Asterisk working great with an X100P, but now I've got up to a TDM400P, with an FXO and FXS card. I've changed what I thought I needed to, and upgraded the Zaptel stuff from 1.0.9 to current CVS. Asterisk is 1.0.9. I'm getting the following when taking the phone off hook: Nov 12 16:23:13 WARNING[3694]: chan_zap.c:5854 handle_init_event: Unable to play dialtone on channel 3 -- Starting simple switch on 'Zap/3-1' -- Hungup 'Zap/3-1' Config files: /etc/asterisk/zapata.conf- [channels]language=encontext=inbound-analogsignalling=fxs_ksusecallerid=yesechocancel=yesechocancelwhenbridged=yeschannel = 4 signalling=fxo_kscontext=defaultlanguage=enchannel = 3 /etc/zaptel.conf- loadzone = ukdefaultzone=us fxsks=4fxoks=3 Is there anything I've missed? Andrew ___ --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] Does IAX2 Trunk Work between IAX and SIP
As far as I know, if the second server has only sip, it is going to be difficult to connect each other. If that is the case, yes it is going to save bandwith. Your total bandwith always it is going to be 0 kbps. Regards, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chawki hammoud Sent: Saturday, November 12, 2005 10:20 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Does IAX2 Trunk Work between IAX and SIP Hi: Does Trunk=yes in IAX save bandwidth as it should in case the other server (voip provider) has SIP only? Regards; Chawki __ 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] Problems after upgrade...
Thanks Tom, That was it, after upgrading the kernel with Yum, it didn't change the link for the modules. Fixed it manually, recompile everything and we are up again. Best regards, Francois On Nov 12, 2005, at 12:11 AM, Francois Meehan wrote: Hi all, I have upgrade my kernel and asterisk to their latest release on a Centos 4.1 box, now it won't start anymore. Have you rebuilt Zaptel against your new kernel? If you upgrade the kernel, you need to rebuild zaptel. 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] Unable to play dialtone
I've had Asterisk working great with an X100P, but now I've got up to a TDM400P, with an FXO and FXS card. I've changed what I thought I needed to, and upgraded the Zaptel stuff from 1.0.9 to current CVS. Asterisk is 1.0.9. I'm getting the following when taking the phone off hook: Nov 12 16:23:13 WARNING[3694]: chan_zap.c:5854 handle_init_event: Unable to play dialtone on channel 3 -- Starting simple switch on 'Zap/3-1' -- Hungup 'Zap/3-1' Config files: /etc/asterisk/zapata.conf- [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 4 signalling=fxo_ks context=default language=en channel = 3 /etc/zaptel.conf- loadzone = uk defaultzone=us fxsks=4 fxoks=3 Is there anything I've missed? Don't know from the limited amount of data that you provided. Here's some things to look at. Make sure you know exactly which modules are fxo and fxs on the TDM card (above says module positions 3 4), and that the definitions in /etc/zaptel.conf match what is installed. An fxo module uses a definition in /etc/zaptel.conf as fxsks=slot. (And, fxs modules use fxoks definitions.) Module slot 4 is the one farest from the TDM card's rear mounting plate that has the four jacks on it. Ensure you are loading wctdm driver (use 'lsmod' to see if its loaded). run 'ztcfg -vv' use 'zttool' to see the TDM card and which modules are recognized. Don't plug the pstn line into the wrong module (rj11 jack) as its likely to blow the module, particularily if ringing voltage from the central office gets dumped into the fxs jack. Use 'zap show status' to understand what asterisk is seeing. ___ --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 play dialtone
Make sure you have a /etc/asterisk/indications.conf Not every method of playing tones requires this, but some do and it's a good idea to have it anyway. Rich Adamson wrote: I've had Asterisk working great with an X100P, but now I've got up to a TDM400P, with an FXO and FXS card. I've changed what I thought I needed to, and upgraded the Zaptel stuff from 1.0.9 to current CVS. Asterisk is 1.0.9. I'm getting the following when taking the phone off hook: Nov 12 16:23:13 WARNING[3694]: chan_zap.c:5854 handle_init_event: Unable to play dialtone on channel 3 -- Starting simple switch on 'Zap/3-1' -- Hungup 'Zap/3-1' Config files: /etc/asterisk/zapata.conf- [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 4 signalling=fxo_ks context=default language=en channel = 3 /etc/zaptel.conf- loadzone = uk defaultzone=us fxsks=4 fxoks=3 Is there anything I've missed? Don't know from the limited amount of data that you provided. Here's some things to look at. Make sure you know exactly which modules are fxo and fxs on the TDM card (above says module positions 3 4), and that the definitions in /etc/zaptel.conf match what is installed. An fxo module uses a definition in /etc/zaptel.conf as fxsks=slot. (And, fxs modules use fxoks definitions.) Module slot 4 is the one farest from the TDM card's rear mounting plate that has the four jacks on it. Ensure you are loading wctdm driver (use 'lsmod' to see if its loaded). run 'ztcfg -vv' use 'zttool' to see the TDM card and which modules are recognized. Don't plug the pstn line into the wrong module (rj11 jack) as its likely to blow the module, particularily if ringing voltage from the central office gets dumped into the fxs jack. Use 'zap show status' to understand what asterisk is seeing. ___ --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 play dialtone
Don't know from the limited amount of data that you provided. Doh :) Ensure you are loading wctdm driver (use 'lsmod' to see if its loaded). run 'ztcfg -vv' That produces: Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 2 channels configured. use 'zttool' to see the TDM card and which modules are recognized. It sees the card, says 2 out of 4 lines/modules are configured Don't plug the pstn line into the wrong module (rj11 jack) as its likely to blow the module, particularily if ringing voltage from the central office gets dumped into the fxs jack. It's definitely in the right socket, as the other channel connected to the phone line still gets answered after x rings, as I had setup on the X100. Use 'zap show status' to understand what asterisk is seeing. That command doesn't seem to exist in my version? Having played some more, I think it's to do with asterisk or perhaps the odd dialplan I have set up - I've modified the extensions.conf file and the attached phone will ring and answer fine! I'll put the default confs back, see what happens and post again if theres still problems.. Andrew ___ --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] vigortalk and transfers
Hi, I have an analog phone connected to a VigorTalk adapter. When I have an active call and press R the call seems to be parked, the other end hears MOH, but how do I transfer the call to another extension. All options in features.conf is commented out, do I need configure something here or is there any other way I can transfer the call? /urban ___ --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] Swissvoice ip20 MGCP issues
Hello there, i have a swissvoice MGCP ip20 as the subject suggests. I have it connecting to asterisk and it shows in `mgcp show endpoints`: *CLI mgcp show endpoints Gateway '192.168.0.10' at 192.168.0.10 (Static) -- 'aaln/[EMAIL PROTECTED] in 'desk-phone' is idle however when trying to dial '5' (should answer and playback hello-world') I get an error tone from the phone and the following messages from asterisk: 1. Non-debug: -- Message check requested for mailbox /folder INBOX but voicemail not loaded. -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down (i then hung up) -- MGCP handle_request(aaln/[EMAIL PROTECTED]) ast_channel already destroyed, resending DLCX. -- MGCP handle_request(aaln/[EMAIL PROTECTED]) set vmwi(-) 2. Debug: MGCP Debugging Enabled MGCP read: NTFY 10099 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 X: 25b013c5 O: hd from 192.168.0.10:2427 Verb: 'NTFY', Identifier: '10099', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 10099 OK to 192.168.0.10:2427 -- Creating connection for aaln/[EMAIL PROTECTED] in cxmode: sendrecv callid: 429ea2c56c9ec4b4 We're at 192.168.0.3 port 21790 Answering with capability 8 Posting Request: CRCX 10 aaln/[EMAIL PROTECTED] MGCP 1.0 C: 429ea2c56c9ec4b4 L: p:20, a:PCMA M: sendrecv X: 6c9ec4b4 v=0 o=root 12521 12521 IN IP4 192.168.0.3 s=session c=IN IP4 192.168.0.3 t=0 0 m=audio 21790 RTP/AVP 8 a=rtpmap:8 PCMA/8000 to 192.168.0.10:2427 -- MGCP Asked to indicate tone: L/dl on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 11 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 25b013c5 R: L/hu(N),L/hf(N),D/[0-9#*](N) S: L/dl to 192.168.0.10:2427 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down MGCP read: 200 10 OK I: 19 v=0 o=- 25 0 IN IP4 192.168.0.10 s=- c=IN IP4 192.168.0.10 b=AS:81 t=0 0 a=sendrecv m=audio 3 RTP/AVP 8 a=ptime:20 from 192.168.0.10:2427 Verb: '200', Identifier: '10', Endpoint: 'OK', Version: '(null)' 2 headers, 9 lines Capabilities: us - 12, them - 8, combined - 8 Non-codec capabilities: us - 1, them - 0, combined - 0 MGCP read: 200 11 OK from 192.168.0.10:2427 Verb: '200', Identifier: '11', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines MGCP read: NTFY 10100 aaln/[EMAIL PROTECTED] MGCP 1.0 NCS 1.0 X: 25b013c5 O: 5 from 192.168.0.10:2427 Verb: 'NTFY', Identifier: '10100', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 10100 OK to 192.168.0.10:2427 -- MGCP Asked to indicate tone: L/dl on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Posting Request: RQNT 12 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 25b013c5 R: L/hu(N),L/hf(N),D/[0-9#*](N) S: L/dl to 192.168.0.10:2427 -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/aaln/[EMAIL PROTECTED] -- MGCP Asked to indicate tone: on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Queueing Request: RQNT 13 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 25b013c5 R: L/hu(N),L/hf(N),D/[0-9#*](N) to 192.168.0.10:2427 -- MGCP mgcp_hangup(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED] -- Delete connection 19 aaln/[EMAIL PROTECTED] with new mode: sendrecv on callid: 429ea2c56c9ec4b4 Posting Request: DLCX 14 aaln/[EMAIL PROTECTED] MGCP 1.0 C: 429ea2c56c9ec4b4 X: 6c9ec4b4 I: 19 to 192.168.0.10:2427 -- MGCP Asked to indicate tone: L/ro on aaln/[EMAIL PROTECTED] in cxmode: sendrecv Queueing Request: RQNT 15 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 25b013c5 R: L/hu(N),L/hf(N),D/[0-9#*](N) S: L/ro to 192.168.0.10:2427 MGCP read: 200 12 OK from 192.168.0.10:2427 Verb: '200', Identifier: '12', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines Posting Queued Request: RQNT 13 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 25b013c5 R: L/hu(N),L/hf(N),D/[0-9#*](N) to 192.168.0.10:2427 MGCP read: 250 14 OK P: PS=128,OS=22016,PR=0,OR=0,PL=0,JI=0,LA=0 from 192.168.0.10:2427 Verb: '250', Identifier: '14', Endpoint: 'OK', Version: '(null)' 2 headers, 0 lines MGCP read: 200 13 OK from 192.168.0.10:2427 Verb: '200', Identifier: '13', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines Posting Queued Request: RQNT 15 aaln/[EMAIL PROTECTED] MGCP 1.0 X: 25b013c5 R: L/hu(N),L/hf(N),D/[0-9#*](N) S: L/ro to 192.168.0.10:2427 MGCP read: 200 15 OK from 192.168.0.10:2427 Verb: '200', Identifier: '15', Endpoint: 'OK', Version: '(null)' 1 headers, 0 lines (hang up here) from 192.168.0.10:2427 Verb: 'NTFY', Identifier: '10101', Endpoint: 'aaln/[EMAIL PROTECTED]', Version: 'MGCP 1.0' 3 headers, 0 lines Handling request 'NTFY' on aaln/[EMAIL PROTECTED] Transmitting: 200 10101 OK to 192.168.0.10:2427 -- MGCP handle_request(aaln/[EMAIL PROTECTED]) ast_channel already destroyed, resending DLCX. -- Delete connection aaln/[EMAIL PROTECTED] with new mode: recvonly on callid: Posting Request: DLCX 16 aaln/[EMAIL
Re: AW: [Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs
[EMAIL PROTECTED] wrote: Thanks Armin, this version is working, but I still have an undefined symbol in another module: [pbx_wilcalu.so]Nov 5 18:51:12 WARNING[11348]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Nov 5 18:51:12 WARNING[11348]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed! Can you also help me on that issue? Thanks and Regards Markus To my knowledge, that module has nothing to do with CAPI. I don't honestly know what it does. (will call you) What I can say is that with 1.2 RC2 (latest from CVS) and chan_capi-cd 0.6 (latest from sourceforge cvs) as of 20:00 12/11/05 GMT on RedHat 9, I get exactly the same error when loading on a freshly sanitised system with all traces of previous asterisk installations removed. HOWEVER, if you add a noload = pbx_wilcalu.so in modules.conf you can make the error go away. (but this is probably a bad thing since I don't know what that module does!) But unfortunately, for me at least, I then end up with errors about: app_capiCD.so app_capiHOLD.so app_capiRETRIEVE.so app_capiECT.so and app_capiMCID.so For example: [app_capiCD.so]Nov 12 20:24:55 WARNING[19197]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber Nov 12 20:24:55 WARNING[19197]: loader.c:554 load_modules: Loading module app_capiCD.so failed! # Ouch ... error while writing audio data: : Broken pipe No matter which of the modules you comment out above, the same thing happens -- the error is always about app_capi_MessageNumber Armin (or anybody) -- have I missed something out/done something wrong, or is it a compatibility issue between chan_capi-cm 0.6 and Asterisk 1.2 RC2? Faris. ___ --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: AW: [Asterisk-Users] chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs
On Sat, 12 Nov 2005, Faris Raouf wrote: [EMAIL PROTECTED] wrote: Thanks Armin, this version is working, but I still have an undefined symbol in another module: [pbx_wilcalu.so]Nov 5 18:51:12 WARNING[11348]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol: ast_pthread_create Nov 5 18:51:12 WARNING[11348]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed! Can you also help me on that issue? Thanks and Regards Markus To my knowledge, that module has nothing to do with CAPI. I don't honestly know what it does. (will call you) What I can say is that with 1.2 RC2 (latest from CVS) and chan_capi-cd 0.6 (latest from sourceforge cvs) as of 20:00 12/11/05 GMT on RedHat 9, I get exactly the same error when loading on a freshly sanitised system with all traces of previous asterisk installations removed. HOWEVER, if you add a noload = pbx_wilcalu.so in modules.conf you can make the error go away. (but this is probably a bad thing since I don't know what that module does!) But unfortunately, for me at least, I then end up with errors about: app_capiCD.so app_capiHOLD.so app_capiRETRIEVE.so app_capiECT.so and app_capiMCID.so For example: [app_capiCD.so]Nov 12 20:24:55 WARNING[19197]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: ast_capi_MessageNumber Nov 12 20:24:55 WARNING[19197]: loader.c:554 load_modules: Loading module app_capiCD.so failed! # Ouch ... error while writing audio data: : Broken pipe No matter which of the modules you comment out above, the same thing happens -- the error is always about app_capi_MessageNumber Armin (or anybody) -- have I missed something out/done something wrong, or is it a compatibility issue between chan_capi-cm 0.6 and Asterisk 1.2 RC2? I cannot tell abything about the pbx_wilcalu.so issue, but with current chan_capi-cm all app_capi* modules are obsolete and may not be used any more. Just have a look at the chan_capi-cm package/cvs-contents, them modules are removed, so why are you trying to load these? As stated in README, the functionality is not part of chan_capi.so itself. Armin ___ --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] Sipura SPA-3000 setup in Brazil
Paul,I have been using a SPA-3000 here in São Paulo, and after some tweaking it is working ok.I do recall that I had to set some line parameters due to the differences, but I need to open the configs of the ATA again Meanwhile, if you have any additional info let me know, I might be able to helpCheers,oZ - Original Message - From: Paul Davidson To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, October 05, 2005 8:25 PM Subject: [Asterisk-Users] Sipura SPA-3000 setup in Brazil All- I'm attempting to set up a Sipura SPA-3000 in Sao Paolo, Brazil. Not being a portuguese speaker, I'm having a rough time of finding the relevant information on how to make the thing pick up the PSTN line and make an outbound call. The sipura in question works fine on a bench connected to a POTS line in the US, but is now plugged in in Sao Paolo. The immediate thing I notice is that the voltage is high by US standards- 66v, as opposed to 48. But no combination of settings will seem to make it work properly- attempted outbound calls generate dead air. A brazilian POTS phone hooked up to the FXS port on the unit works just fine- I can ring it, and it can call into Asterisk. Am I barking up the wrong tree, and does the device have to be different for it to work in Brazil, or is there some magic I need to apply to the Sipura config to make it work? Any previous experience shared would be welcomed, and eventually, published on the wiki. -pbd ___ --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] Echo
I've got a customer on an IAXy and another with their own Asterisk box as a PBX with an array of Cisco, GrandStream, ATCOM, and xten hard\soft phones. Same LEC, same Asterisk box on our end, same broadband provider on the client ends With no packet loss, 15 ms pings, 13 hops, the IAXy sometimes has an echo, some times not. The client with the Asterisk box... no problems at all. What could I do to figure out what's going on here? --Mike This message was sent using IMP, the Internet Messaging Program. ___ --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 with this
I'm trying to get this to work, but it always goes to step 4 - there
something I don't understand about LEN with GotoIf:
exten = _,1,NoOp,${CALLERIDNUM} ; CID as
received
exten = _,2,GotoIf([LEN(${CALLERIDNUM}) = 10]?4:3) ; if CID length =
10 then do nothing
exten = _,3,SetCallerID(${CALLERIDNUM:2}) ; Remove
the first two digits
exten = _,4,NoOp,${CALLERIDNUM} ; CID
after fix
exten = _,5,goto(ext-did,${EXTEN},1)
TIA
Bart
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Re: [Asterisk-Users] Help with this
I think you have to swap the 4 and 3 around in your gotoif - it's true then
false...
PaulH
- Original Message -
From: Bart Fisher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, November 13, 2005 8:32 AM
Subject: [Asterisk-Users] Help with this
I'm trying to get this to work, but it always goes to step 4 - there
something I don't understand about LEN with GotoIf:
exten = _,1,NoOp,${CALLERIDNUM} ; CID
as
received
exten = _,2,GotoIf([LEN(${CALLERIDNUM}) = 10]?4:3) ; if CID length =
10 then do nothing
exten = _,3,SetCallerID(${CALLERIDNUM:2}) ; Remove
the first two digits
exten = _,4,NoOp,${CALLERIDNUM} ; CID
after fix
exten = _,5,goto(ext-did,${EXTEN},1)
TIA
Bart
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Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline
At least the soft limit is explicitly published (X Minutes) as opposed to most companies' policy of There is a soft limit, and we will not tell you what it is, but if you reach or exceed it we will [charge you $100/day | terminate your service | switch you to a more expensive plan without notice]. I am sort-of searching for a new SIP/IAX trunk provider and I would much rather have a policy like Teliax's than the others. -Rusty On 11/12/05, Dane Reugger [EMAIL PROTECTED] wrote: Teliax looks good - not comfortable with the soft limits but love thefree setup![EMAIL PROTECTED] wrote:Have you looked into teliax?4 simultaneous calls on a bus plan is pretty good for less than $50/mo. And I cannot complain about thequality or the support.Greg-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Saul DiazSent: Friday, November 11, 2005 4:34 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Quantumvoice vs Broadvoice - MultilineJulio Arruda wrote:I was testing Broadvoice few weeks before Hurricane Wilma here in FL. Since then, I had been since the landline (Bellsouth), and I had to'remote callfwd' the BS # to my broadvoice #.So, from my impression, is ok for my needs (I got a weird no ringback problem that I kind of solved with a Background trick), and nosurprises yet regarding the bill (my mother in law call Brazil a lotfrom my house, no, she is not aware of the 'unlimited' plan. So I may be in for a surprise in a couple of months).I've no tried several calls at the same time, you may want to askthem..PS: I'm running Asterisk 1.0.9Dane Reugger wrote:We are considering Quantumvoice as a provider -They are telling us they will give us 1 line number but we can have 5 concurrent incoming and outgoing line numbers. Charge is about $45 +extras - this seems considerable less expensive than the competition which seem to focus on.My second choice is BroadVoice $29.99 + $9.99 per additional line (instate only?) - more expensive, less features, and they don't seem loved by many ?Is anyone else using Quantum Voice?It was mentioned earlier that it requires an ATA connection andAsterisk support/compatibility is sketchy at best - I've contacted BV and they responded saying they need 24hrs to look into it?Seems like a popular topic but I'm looking for 2-3 lines - I only need one number but need to be able to make or receive several callsat a time?Any advice or recommendations appreciated - I want to portmy number but I'm running out of time and must make a decision very soon.Thanks,Dane ReuggerCrescent City Technologies New Orleans, LA 70112___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://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 Broadvoice only allows only the normal 3 way calling so is 2 channelsfor #about BV i got a lot of water under the bridge every works ok supperok for times. then BV brokes without you make a single change in your asterisk server and stop working.. if u call support you are the guywith the problem.. yes BV support sucks, and it took me 9 phone calls,12 emails, 3 chargeback and 2 call to my bank to remove myself from their billing all them well documented...so my advice nothing can be worts than BV.regardsSaul___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://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-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://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.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Does IAX2 Trunk Work between IAX and SIP
No. IAX and SIP are two completely different protocols for sending voice across IP networks. IAX-Trunking is a feature of IAX, and the SIP protocol does not have any such method for conserving bandwidth by combining data from multiple calls into one packet. -Rusty On 11/12/05, chawki hammoud [EMAIL PROTECTED] wrote: Hi:Does Trunk=yes in IAX save bandwidth as it should incase the other server (voip provider) has SIP only?Regards;Chawki__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.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] Warning CONFIG_ZAPATA_DEBUG on 2.6.14
Hi Upgraded to Gentoo 2.6.14-r2. When compiling zaptel, warning appears. Zaptel module loads fine. Cannot remember seeing this on 2.6.13. Is there another Kernel switch that needs to set. CRC and RTC is set in kernel. make[1]: Entering directory `/usr/src/linux-2.6.14-gentoo-r2' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:1736:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:1923:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:3032:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:3039:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:3048:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:3295:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:5287:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:5806:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:5876:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:5899:5: warning: CONFIG_ZAPATA_DEBUG is not defined /usr/src/zaptel/zaptel.c:176: warning: 'fcstab' defined but not used Master ___ --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] REaltime does not unregister sip peers on the fly but not only...
In addition to the mail below, It is not the realtime! ARA is great. Moving the peers to sip.conf, and ignoring extconfig.conf for a test, discovered that, when left empty (secret=blank_space) is ignored as commented (;secret=whatever). Obviously the sip channel was actually prepared for the realtime sip_buddies table. Means, columns secretmd5secret were left empty are to be considered Not set. Thinking out load, for me, secret=blank_space meant that either the client should have literally blank password or should not be able to register, isn't it. If you don't want secrets you comment it like this (;secret=not_needed). However, do not leave secret empty if you require passwords from your users. Simply set secret=/ or something similar :). On Saturday 12 November 2005 17:45, [EMAIL PROTECTED] wrote: Hi, chick*CLI show version files chan_sip.c File Revision chan_sip.cRevision: 1.907 chick*CLI show version files pbx_realtime.c File Revision pbx_realtime.cRevision: 1.15 chick*CLI show version Asterisk CVS HEAD built by root @ chick on a i686 running Linux on 2005-11-09 14:28:18 UTC extconfig.conf sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies extentions = mysql,asterisk,extensions_table Registered sip friends work great through realtime. However, setting in sip.conf: rtcachefriends=yes rtupdate=yes rtautoclear=yes ignoreregexpire=no ( trying to have show peers available, but attempting to clear the cache) and deleting the secret the friend , this friend :) stays registered: Seeding..., Saved... and all. Weren't they supposed with realtime to get busted on the fly at the next registration without sip reload? Or just created (on the fly)? Anybody? benchev ___ --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] NEC NEAX 2400 Integration with Asterisk
FYI, I was able to get my NEC NEAX 2400 and my * box to talk to each other. For those who want to know more Ive added a wiki page: http://www.voip-info.org/wiki/view/Asterisk+NEAX2400 -MC ___ --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 Buddy Feature
I am having the same problems. The polycom phonesthe 501 or 601 or 301 will list more more than 7 buddies neither will the 601 with an expansion module monitor more than 7 other phones. Is there anyone out there who can explain waht is happening. My reseller can not help. I am surprised no one has reported the reason for the problem or or even a word from the manafacturer. Please someone our ther help. the phones are great but this is a big issue maraba ___ --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] Example of Pass-Thru Codec
Hello All, Can anyone give me an example of how can I configure Asterisk to Pass- Thru G729 and G723.1 codec? Thanks, Neal ___ --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] WARNING[3035]: Invalid priority/label ' ' at line 17
I just recently upgraded to the latest HEAD, and am now getting the following warning: -- Including context 'fromcnet' in context 'pots' Nov 12 18:45:17 WARNING[3035]: pbx_config.c:1697 pbx_load_module: Invalid priority/label '' at line 17 -- Including context 'longdistance' in context 'international' I have added a comment line above and below every config file that I have in /etc/asterisk, and the warning never changes. What's up with this? And will it affect anything? TIA ___ --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] How to let caller continue after Dial cmd
This might help you:
${GOTO_ON_BLINDXFR} Transfer to the specified context/extension/priority
after a blind transfer (use ^ characters in place of
| to separate context/extension/priority when setting
this variable from the dialplan)
Check /usr/src/asterisk/doc/README.variables
On 11/12/05, George Pajari [EMAIL PROTECTED] wrote:
We have a need to allow the caller who is in the middle of a call (i.e.
who is already bridged between a PRI channel and a SIP channel as the
result of entering a Dial cmd in the current context) to type something
like ## to cause the called party to be disconnected and to return
from the Dial command with a distinctive STATUS so they can proceed to
do other things within the context. Sort of a combination of the H and g
flags to the Dial cmd.
Any thoughts about whether or not this is possible in 1.0 or 1.2?
If not, is there sufficient interest in such a feature for us to submit
it once complete (I don't want to go through the effort to properly
document, post, and maintain such a patch if its an improvement no one
wants).
--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102)
www.netvoice.ca www.ip-centrex.ca
www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
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Re: [Asterisk-Users] Snom 360 Opinions
Curren, Can you tell us a little more about the environment you are deploying these phones in, how many phones and what kind of setup? Omar A. SabekOn 11/12/05, Curren C. Calhoun [EMAIL PROTECTED] wrote: Thanks that's the type of info I'm looking for...I've heard some early grumblings but wanted to see if anything else has comeup... From: Remco Barende [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat, 12 Nov 2005 09:50:01 +0100 (CET) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Snom 360 Opinions I'm not too pleased with the phones, I have about 40 of them, some of the displays tend to die and the dial pad feels to 'mushy' IMHO, just like the keys on a good old ZX80 computer Also I'm having some issues with sound quality on some phones, but I still need to switch some phones to see if that is really an issue of the phone. Also if you want to use * call files, with the 360 you will run into a big where the call is being redialled as if it failed while in fact the call is ongoing. Annoying and haven't found out if that is an * bug or Snom bug. The Snom 190's do not have this problem. Just my $0.02 (which is really not a lot these days!) :) On Sat, 12 Nov 2005, Curren C. Calhoun wrote: I¹m looking to add in some Snom 360 phones, could anyone give thoughts or opinions about the speakerphone, general quality... Also the phone would need to be powered over Ethernet... I like some of the listed features and the expandability of the phone but am open to any other suggestions as well... Thanks Curren___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://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.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 REGISTER
From the dump that I have attached It looks like the first attempt at register does not work then followd by a second register which then works. This is happening on all the SIP phone attach to asterisk. The version of asterisk here is 1.2.0b2. Here is sip.conf for ext 204 [204] username=204 type=friend secret= md5secret=356381525bb1969a32743b58db400342 auth=md5 record_out=Adhoc record_in=Adhoc port=5060 [EMAIL PROTECTED] host=dynamic context=from-sip canreinvite=no callerid=Library 204 Is there somethng that I am missing to have phone only reigster once and not get the 401 unauthorized on the first attempt which then get follow by the same register but get 200 Ok. No. TimeSourceDestination Protocol Info 3 0.494325192.168.3.70 192.168.3.28 SIP Request: REGISTER sip:192.168.3.28 Frame 3 (675 bytes on wire, 675 bytes captured) Ethernet II, Src: 192.168.3.70 (00:0e:08:ca:5f:2d), Dst: 192.168.3.28 (00:a0:c9:e7:9c:6e) Internet Protocol, Src: 192.168.3.70 (192.168.3.70), Dst: 192.168.3.28 (192.168.3.28) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:192.168.3.28 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.3.70:5060;branch=z9hG4bK-8c701711 From: Great Room sip:[EMAIL PROTECTED];tag=51baed127acfd9a6o0 SIP Display info: Great Room SIP from address: sip:[EMAIL PROTECTED] SIP tag: 51baed127acfd9a6o0 To: Great Room sip:[EMAIL PROTECTED] SIP Display info: Great Room SIP to address: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 79 REGISTER Max-Forwards: 70 Authorization: Digest username=204,realm=asterisk,nonce=6948ea10,uri=sip:192.168.3.28,algorithm=MD5,response=3c05b1dab053a739d7c8ad941ac98cee Contact: Great Room sip:[EMAIL PROTECTED]:5060;expires=60 Contact Binding: Great Room sip:[EMAIL PROTECTED]:5060;expires=60 User-Agent: Sipura/SPA3000-3.1.7(GWg) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura No. TimeSourceDestination Protocol Info 4 0.495379192.168.3.28 192.168.3.70 SIP Status: 100 Trying(1 bindings) Frame 4 (473 bytes on wire, 473 bytes captured) Ethernet II, Src: 192.168.3.28 (00:a0:c9:e7:9c:6e), Dst: 192.168.3.70 (00:0e:08:ca:5f:2d) Internet Protocol, Src: 192.168.3.28 (192.168.3.28), Dst: 192.168.3.70 (192.168.3.70) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 192.168.3.70:5060;branch=z9hG4bK-8c701711;received=192.168.3.70 From: Great Room sip:[EMAIL PROTECTED];tag=51baed127acfd9a6o0 SIP Display info: Great Room SIP from address: sip:[EMAIL PROTECTED] SIP tag: 51baed127acfd9a6o0 To: Great Room sip:[EMAIL PROTECTED] SIP Display info: Great Room SIP to address: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 79 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Contact Binding: sip:[EMAIL PROTECTED] Content-Length: 0 No. TimeSourceDestination Protocol Info 5 0.495662192.168.3.28 192.168.3.70 SIP Status: 401 Unauthorized(1 bindings) Frame 5 (555 bytes on wire, 555 bytes captured) Ethernet II, Src: 192.168.3.28 (00:a0:c9:e7:9c:6e), Dst: 192.168.3.70 (00:0e:08:ca:5f:2d) Internet Protocol, Src: 192.168.3.28 (192.168.3.28), Dst: 192.168.3.70 (192.168.3.70) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Status-Line: SIP/2.0 401 Unauthorized Message Header Via: SIP/2.0/UDP 192.168.3.70:5060;branch=z9hG4bK-8c701711;received=192.168.3.70 From: Great Room sip:[EMAIL PROTECTED];tag=51baed127acfd9a6o0 SIP Display info: Great Room SIP from address: sip:[EMAIL PROTECTED] SIP tag: 51baed127acfd9a6o0 To: Great Room sip:[EMAIL PROTECTED];tag=as6a7c3473 SIP Display info: Great Room SIP to address: sip:[EMAIL PROTECTED] SIP tag: as6a7c3473 Call-ID: [EMAIL PROTECTED] CSeq: 79 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Contact Binding: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=3dbc57fe Content-Length: 0 No. TimeSource
Re: [Asterisk-Users] Does IAX2 Trunk Work between IAX and SIP
Carlos Alperin wrote: As far as I know, if the second server has only sip, it is going to be difficult to connect each other. If that is the case, yes it is going to save bandwith. Your total bandwith always it is going to be 0 kbps. Hehe funny!!! -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) 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] REaltime does not unregister sip peers on the fly but not only...
[EMAIL PROTECTED] wrote: In addition to the mail below, It is not the realtime! ARA is great. Moving the peers to sip.conf, and ignoring extconfig.conf for a test, discovered that, when left empty (secret=blank_space) is ignored as commented (;secret=whatever). Obviously the sip channel was actually prepared for the realtime sip_buddies table. Means, columns secretmd5secret were left empty are to be considered Not set. If you don't have something for secret in the sip.conf file its exactly the same. Thinking out load, for me, secret=blank_space meant that either the client should have literally blank password or should not be able to register, Why would you want to stop someone from registering? isn't it. If you don't want secrets you comment it like this (;secret=not_needed). However, do not leave secret empty if you require passwords from your users. Simply set secret=/ or something similar :). Eh?! You're weird! :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) 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: SV: [Asterisk-Users] Call p2p
Amund Nygaard wrote: Do you know anywhere to find information about this? set the option canreinvite=yes in the sip.conf section for that user and make sure you don't have anything in the dial line that would keep asterisk in the communication (i.e. t or T etc) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) 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] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 18
PLEASE DO NOT POST IN HTML! :) Gervais de Montbrun wrote: YPE HTML PUBLIC =22-//W3C//DTD HTML 4.0 Transitional//EN=22 htmlheadmeta http-equiv=3D=22Content-Type=22 content=3D=22text/html; c= harset=3DISO-8859-1=22 style type=3D=22text/css=22body=7Bmargin-left:10px;margin-right:10px;marg= in-top:10px;margin-bottom:10px;=7D/style /head body marginleft=3D=2210=22 marginright=3D=2210=22 margintop=3D=2210=22 mar= ginbottom=3D=2210=22 font face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=2300=22 style=3D= =22font-family:Geneva;font-size:10pt;color:=2300;=22bAsterisk Users = Mailing List - Non-Commercial Discussion lt;a href=3D=22mailto:asterisk-u= sers=40lists.digium.com=22asterisk-users=40lists.digium.com/agt; on Thu= rsday, November 10, 2005 at 5:16 AM -0400 wrote:br /b/fontspan style=3D=22background-color:=23d0d0d0=22font face=3D=22G= eneva=22 size=3D=22+0=22 color=3D=22=2300=22 style=3D=22font-family:Gen= eva;font-size:12pt;color:=2300;=22the 12SP should work/font/spanf= ont face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=2300=22 style=3D= =22font-family:Geneva;font-size:12pt;color:=2300;=22br /fontspan style=3D=22background-color:=23d0d0d0=22font face=3D=22Genev= a=22 size=3D=22+0=22 color=3D=22=2300=22 style=3D=22font-family:Geneva;= font-size:12pt;color:=2300;=22br Sergiobr /font/spanfont face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=230= 0=22 style=3D=22font-family:Geneva;font-size:12pt;color:=2300;=22br I half-managed to get my 12SP working with sccp and I am able to call it wi= th my ATA. The ATA and my cordless phone is still configured using SIP.br br I can call out from my Cisco 12 SP+ and everything seems to be working fine= . I can not however receive calls on the 12SP. The phone rings and it can b= e answered, but there is no audio at all. When I hang up, I can see that th= e phone reset. Also if I call in on the PSTN, I get similar results except = even after I hang up my 12SP the Zap channel is not released. It stayed tha= t way for at least 1 minute after hanging up until I restarted asteriskbr br What am I doing wrong?br br I'm running rc-1 of asterisk with the latest sccp 20051108.br br Thanks in advance,br Gervaisbr ---br br /etc/asterisk/sccp.confbr =5Bgeneral=5Dbr keepalive =3D 5 nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;br context =3D defaultbr dateFormat =3D D.M.Y nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;= nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nb= sp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;= nbsp;nbsp;nbsp;;=23160;M-D-Y=23160;in=23160;any=23160;order=23160;(= 5=23160;chars=23160;max)br bindaddr =3D 192.168.1.125 nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;= nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nb= sp;nbsp;nbsp;nbsp;nbsp;nbsp;=23160; ;=23160;asterisk=23160;box.br port =3D 2000 nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;= nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbs= p;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;= nbsp;nbsp;nbsp;nbsp;nbsp;=23160;; listen=23160;on=23160;port=23160;= 2000=23160;(Skinny,=23160;default)br debug =3D 0br br =5Bdevices=5Dbr type nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;nbsp;=3D 12br description =3D Officebr tzoffset nbsp;nbsp;nbsp;=3D 0br autologin nbsp;nbsp;=3D 140br speeddial nbsp;nbsp;=3D 500,500,500=40defaultbr device =3Dgt; SEP003080629796br br br =5Blines=5Dbr id =3D 140br pin =3D 1234br label =3D quot;TLS Groupquot;br description =3D Officebr context =3D defaultbr callwaiting =3D 1br incominglimit =3D 2br mailbox =3D 1000br vmnum =3D *98br cid_name =3D Officebr cid_num =3D 140br line =3Dgt; 140br br /etc/asterisk/sip.confbr =5Bgeneral=5Dbr port =3D 5060br bindaddr =3D 0.0.0.0br context =3D defaultbr br disallow=3Dallbr allow=3Dg729br allow=3Dgsmbr allow=3Dspeexbr allow=3Dilbcbr br =5B500=5Dbr type=3Dfriendbr username=3D500br callerid=3Dquot;TLS Groupquot;br secret=3Dmypasswordbr canreinvite=3Dnobr host=3Ddynamicbr dtmfmode=3Drfc2833br mailbox=3D1000br nat=3D1br br /etc/asterisk/extensions.confbr exten =3Dgt; 140,1,Dial(SCCP/140,20,tr)br exten =3Dgt; 140,2,Voicemail(u140)br exten =3Dgt; 140,3,Goto(mainmenu,s,2)br exten =3Dgt; 140,102,Voicemail(b140)br exten =3Dgt; 140,103,Goto(mainmenu,s,2)br br /fontfont face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=23DD=22 st= yle=3D=22font-family:Geneva;font-size:12pt;color:=23DD;=22This is what= is displayed in the console when I try to call the 12SP from the ATAbr /fontfont face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=2300=22 st= yle=3D=22font-family:Geneva;font-size:12pt;color:=2300;=22nbsp;nbsp;= nbsp;-- Executing Dial(quot;SIP/500-fc17quot;, quot;SCCP/140=7C20=7Ctr= quot;) in new stackbr nbsp;nbsp;nbsp;nbsp;-- Called 140br nbsp;nbsp;nbsp;nbsp;-- SCCP/140-0001 is ringingbr nbsp;nbsp;nbsp;nbsp;-- SCCP/140-0001 answered SIP/500-fc17br Nov 10 22:06:05 WARNING=5B1693=5D: sccp_socket.c:308 sccp_socket_thread: SE=
Re: [Asterisk-Users] Possible problem with Zaptel/Asterisk with 1.2rc1
Waldo Rubinstein wrote: I upgraded one of our gateways connected to the PSTN with a TE410P to 1.2rc1. Any ideas? Maybe change DEFAULT_CID_RINGS in /usr/src/asterisk/channels/chan_zap.c -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) 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] RE: (BAD!!!) Sound quality of the NEW GRANDSTREAM BT 101 and 102 MODEL
Please post in PlainText not HTML to this list. Thanks! :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) 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] Can't create iax channel
Jason Walker wrote: The statement of zaptel being required is strange...I use IX trunking exclusively for my servers. Two of them have no zaptel/Digium hardware and the trunk calls are fine. IAX trunks require a zaptel timing source, be it hardware or ztdummy. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) 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] Play message and dial extensions simultaneously
Hugh Jackman wrote: How that could possibly be done with a special class of MOH as the file will continue to play from wherever it stops last time? Can we force * to spawn a MOH process for every incoming call? Raw music on hold will do this. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) 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] txfax and rxfax problem with spandsp 0.0.2pre21c and 1.2rc1
I am getting the same with cvs head as of today. on Thu, 10 Nov 2005 15:29:05 -0600 Anton Krall [EMAIL PROTECTED] wrote: Hi Steve! I tried compiling the recent spandsp 0.0.2pre21c with 1.2rc1 and I also compiled unicall for r2mfc support. R2mfc seems to be working great! But when I load everything up I get this on the CLI: Nov 10 15:28:49 WARNING[26895]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler And rxfax/txfax doesn't load and coredumps asterisk. Any ideas? Is there any relation or problem if you use unicall and txfax at the same time? ___ --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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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] A2billing with Mysql-5.0.15
Hi
I was using a2billing with mysql-4.1.12 and php-5.0.4 very successfully
(thanks to areski for this great project and its invaluable assistance
to solve some issues in my last installation...) now I´ve upgraded
mysql to last release 5.0.15 and, without changes in 'mya2billing'
database I am able to make calls, create and see created cards,
etc, but I get this errors when invoke CDR´s in both admin or
user interfase:
Database error: Invalid SQL: SELECT t1.starttime, t1.src,
t1.calledstation, t1.destination, t1.sessiontime, t1.username,
t1.terminatecause, t1.sipiax, t1.calledrate, t1.sessionbill FROM call
t1 WHERE UNIX_TIMESTAMP(t1.starttime) =
UNIX_TIMESTAMP('2005-11-12') ORDER BY t1.starttime DESC LIMIT 0,25
MySQL Error: 1064 (You have an error in your SQL syntax; check
the manual that corresponds to your MySQL server version for the right
syntax to use near 'call t1 WHERE UNIX_TIMESTAMP(t1.starttime) =
UNIX_TIMESTAMP('2005-11-12') ORDE' at line 1)
Database error: Invalid SQL: SELECT substring(t1.starttime,1,10)
AS day, sum(t1.sessiontime) AS calltime, sum(t1.sessionbill) AS cost,
count(*) as nbcall FROM call t1 WHERE UNIX_TIMESTAMP(t1.starttime)
= UNIX_TIMESTAMP('2005-11-12') GROUP BY
substring(t1.starttime,1,10) ORDER BY day
MySQL Error: 1064 (You have an error in your SQL syntax; check
the manual that corresponds to your MySQL server version for the right
syntax to use near 'call t1 WHERE UNIX_TIMESTAMP(t1.starttime) =
UNIX_TIMESTAMP('2005-11-12') GROU' at line 1)
Database error: Invalid SQL: SELECT count(*) FROM call t1 WHERE UNIX_TIMESTAMP(t1.starttime) = UNIX_TIMESTAMP('2005-11-12')
MySQL Error: 1064 (You have an error in your SQL syntax; check
the manual that corresponds to your MySQL server version for the right
syntax to use near 'call t1 WHERE UNIX_TIMESTAMP(t1.starttime) =
UNIX_TIMESTAMP('2005-11-12')' at line 1)
Database error: next_record called with no query pending.
MySQL Error: 1064 (You have an error in your SQL syntax; check
the manual that corresponds to your MySQL server version for the right
syntax to use near 'call t1 WHERE UNIX_TIMESTAMP(t1.starttime) =
UNIX_TIMESTAMP('2005-11-12')' at line 1)
some advice?
rafael
-- rrgv
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Re: [Asterisk-Users] MFC/R2
It would have been better to send such a long log directly to me, rather than to the mailing list. Ok . That said, the log did the job. I found the problem. I just posted another update to the MFC/R2 software - 0.0.2e and 0.0.3pre8 Regards, Steve I installed the new libmfcr2 today. After that, I did a test with it and the result was very good, but I will only have a complete test, with a continous load of calls, next monday so I could send you one more detailed result. Thank you for your attention and for your hard work. I hope you continue having success in yours projects. Maybe, if you need any information about Brazil telephony standards I could help you. Best regards, Loureiro. ___ --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 1.2.0-RC1 Crashing with g729 codec and ATA 1
Matt Riddell on November 12, 2005 at 9:53 PM -0400 wrote: PLEASE DO NOT POST IN HTML! :) Sorry Matt, this is controlled server side for me. The server should be sending in html and plain text and displaying what your email client should be able to read... Isn't this what is happening? Any ideas with my issue? I am currently at the point where I switched to the SCCP protocol for my Cisco 12 SP+ as suggested by Sergio. Things seem to work, but I can not call into my Cisco phone. It rings, but then there is no audio and the phone resets after a short while. Cheers, Gervais ___ --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] codec error connecting to cisco gateway
I hope someone can help me with this I have a cisco gateway trying to dial my box via H.323. The call comes through ok and gets routed properly.. only thing is NO AUDIO I am confident that I have narrowed down the problem to a codec issue. I have the relevant G729 licences which were purchased from diguim. The calling party was calling using G729. I asked them which version of the codec and they told me G729r8 . what ever that means? I get about 150 of these lines in the log file per second while the call is active. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame. Another party with a quantum device tried to call me and it was the same error.. they even tried a SIP call and the result was the same. I had one party call me through a softswitch and both sip and H323 worked fine for that person. In my mind it seem that there is some kind of incompatibility with the audio codec the quintum and cisco are sending and the one asterisk is using. Could this be the case and if so is there a work around? Has anyone else had this issue before or does anyone know possibly if I am wrong what the problem may be? Thanks in advance ___ --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] debian sarge zaptel 1.2 TDM400P
On Sat, Nov 12, 2005 at 03:35:43PM +0800, Dulmandakh Sukhbaatar wrote: I run debian sarge with kernel-image-2.6.8-2-686, compiled and installed zaptel 1.2rc2 without any problem. Modules can be loaded without problem. Also I have TDM04M or TDM card with 4 FXO modules. /etc/zaptel.conf contains, as mentioned in Asterisk. The Future of telephony: fxsks=1-4 loadzone=us defaultzone=us ztcfg -vv shows me: Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. And connected my phone line to the TDM card, of cource RJ45. In Asterisk book: Now that the FXO channel is configured, let’s test it. Run the zttool application and connect your PSTN line to the FXO port on your TDM400P. Once you have a phone line connected to your FXO port, you can watch the card come out of a RED alarm. But nothing becomes red. Any suggestions? Please help and sorry for my poor english. Hmmm... Any problems? Anything not working? Can you dial through the FXO line from Asterisk? -- 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: BUS Error in SPARC/Linux (debian)
On Fri, Nov 11, 2005 at 11:08:25AM +0800, Ryan Pagquil wrote: Hi, I successfully installed asterisk in SPARC64/Linux as the voicemail for my SER installation. No problem when I run it, but the problem is when I forward the voicemail traffic to it, on the user agent side I heard the start of the voice prompt but immediately stopped. I then checked on the server and it says Bus error. What can I do to fix this? Which version of Asterisk? of Debian? If this is from an official deb. maybe you should use reportbug(1). -- 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] 2 SIP phones on Y data connector on 1 ethernet
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Saturday, 12 November 2005 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] 2 SIP phones on Y data connector on 1 ethernet At least use a hub or switch (preferred) But if you MUST use a Y connector make sure the adapter meets the International Data 10T Standard Darn! The ID10T standard! Wow! Hehe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of A_ Navone Sent: Friday, November 11, 2005 3:54 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 2 SIP phones on Y data connector on 1 ethernet 2 SIP phones on Y data connector on 1 ethernet - will that cause problems ? thx in advance _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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] A2billing with Mysql-5.0.15
Rafael R. GV wrote:
Hi
I was using a2billing with mysql-4.1.12 and php-5.0.4 very successfully
(thanks to areski for this great project and its invaluable assistance
to solve some issues in my last installation...) now I´ve upgraded mysql
to last release 5.0.15 and, without changes in 'mya2billing' database I
am able to make calls, create and see created cards, etc, but I get
this errors when invoke CDR´s in both admin or user interfase:
*Database error:* Invalid SQL: SELECT t1.starttime, t1.src,
t1.calledstation, t1.destination, t1.sessiontime, t1.username,
t1.terminatecause, t1.sipiax, t1.calledrate, t1.sessionbill FROM call t1
WHERE UNIX_TIMESTAMP(t1.starttime) = UNIX_TIMESTAMP('2005-11-12') ORDER
BY t1.starttime DESC LIMIT 0,25
MySQL 5.0.15 introduces stored functions and procedures that are invoked
by 'call()'. A2Billing uses 'call' for the name of the cdr table. Find
all occurencies of 'call t1' or ' call ' in A2Billing's sql queries and
replace them to 'calls t1' and ' calls '. Don't forget to rename 'call'
table to 'calls'. In short, latest A2Billing doesn't work on mysql
5.0.15 / PHP5 out of box.
On the side note, MySQL doesn't support more than 1 entry with default
value of DEFAULT now() NOT NULL in one table.
regards,
Vahan
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Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards
Mr. Fleming: Thank you so much for your email -- that's the best news I've heard all week. Kevin P. Fleming wrote: George Pajari wrote: I am aware of all of the above but does the fact you are posting this mean that Digium is now aware of this? It means I am :-) I have been trying to teach the tech support team about this, but I don't think it has quite sunk in yet. I believe that means it's time for another training session... I have had a Digium tech ssh to a machine, run an lspci -vb, report your interrupts are shared and refuse to work on the case until the lspci -vb showed unique IRQs. I understand. Does your post mean that this policy has now changed at Digium? It is in the process of changing, yes. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --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
