Re: [asterisk-users] vzaphfc?
On Wed, Jan 03, 2007 at 07:16:06AM +0100, Remco Barendse wrote: On Wed, 3 Jan 2007, Tzafrir Cohen wrote: P[ 0] -- mISDN Channel Driver Registred -- (BE AWARE THIS DRIVER IS EXPERIMENTAL!) ..Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10603 setup_zap: Unknown signalling method 'bri_cpe_ptmp' our Asterisk is not bristuffed. And you don't expect to use ZapBRI, anyway. BTW: with 1.4 and latest 1.2, bristuffed zaptel could basicaly work with the signalling type pri_cpe/pri_net, though this is not well-tested and may not perform as well as bristuffed asterisk/libpri. Jan 2 23:07:23 ERROR[25747]: chan_zap.c:10228 setup_zap: Signalling must be specified before any channels are. Jan 2 23:07:23 WARNING[25747]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 2 23:07:23 WARNING[25747]: loader.c:554 load_modules: Loading module chan_zap.so failed! You have zaptel channels configured in your zapata.conf . So I should leave both zaptel.conf and zapata.conf completely empty? /etc/zaptel.conf is used to configure the kernel modules. It is used by ztcfg. /etc/asterisk/zapata.conf is used by Asterisk's chan_zap . Asterisk will try to use any channel listed there. If no channel is listed there, Asterisk will not attempt to use any. (except possibly the pseudo channel for zaptel timing, if you happen to have any zaptel device that provides timing, or ztdummy) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [A*UG] How to show a debugging remark in a sip or extensions context?
Thanks very much Chris. I found usage for NoOp and verbose in Future of Telephony Appendix C and it looks like they will do exactly what I need. Larry Chris Tooley wrote: If you mean in the dialplan, you can use NoOp or verbose (verbose being something that will get logged too), and if you mean in the asterisk code, there are logging examples all over the place. -Original Message- From: [EMAIL PROTECTED] on behalf of Larry Alkoff Sent: Tue 1/2/2007 11:22 AM To: Asterisk-users; Austin-asterisk-users Subject: [A*UG] How to show a debugging remark in a sip or extensions context? I would like to show a remark that would show call progress and appear on the CLI screen. The remark should be in the code of a sip [channel] or extentions [context] If I can't send my own remark, what little used 'show' command could I insert in the code? Can this be done? -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM loses server registration
Hello to all When my SNOM (300 or 320) loses Internet connectivity, it loses its Asterisk registration (ok, thats normal). But when the phone is back online, he doesn't try to register in Asterisk. I believe this happens to avoid flooding the private LANs when the Internet link is lost but the problem is that the phones don't try to re-register in the future Sometimes it stays 2 hours without registering to Asterisk. When this happens, the only solution is to reboot it (and hear the users complains) :( How can I avoid this? How can I reduce the time to re-register in SNOM 300 or 320 ? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dubai Caller ID
Hi! I'm trying to set up an asterisk based PBX with a TDM400P +2 FXS +2 FXO modules in UAE/Dubai for home switching / voicemailing. I am using the card Asterisk/Zaptel 1.4.0. I want to include a special route when a certain caller calls into via PSTN. The problem is that I cannot detect the Caller ID. I tryed various setting (cidsignalling, cidstart) in my zapata.conf, here is the last version of it: group=1 signalling=fxs_ks usecallerid=yes cidsignalling=dtfm cidstart=ring hidecallerid=no callerid=asreceived language=en context=zap-incoming channel = 1-2 If you know how to aquire UAE Caller ID with this hardware, please help me. Cheers, Mischi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] yet another faxing issue (outbound only, via ATA)
Bill Gibbs wrote: My next step is to connect the fax machine to a Wildcard X100P. Check to see if there is Echo cancellation in the SPA-1001, and if so turn it off. If there is an adaptive Jitter buffer on the SPA-1001, try changing it to a fixed one (probably no more than 40ms). Why would you connect a fax machine to an X100P, aren't they FXO cards? Have you tried terminating to a VOIP provider? (to see if the problem is with the ATA). Here I use a fax machine connected to a CS6220 which is connected to the asterisk box and terminates with a TDM400P card (so a completely different arrangement). Any other suggestions? Black magic? Voodoo? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] yet another faxing issue (outbound only, via ATA)
Hi all, I was having a similar issue, using TE110P from Digium all incoming faxes were detected and correctly received. When trying to send outbound faxes, they all get broken... I do believe it may be related with Echo Cancel enabled on my Zapata.conf, any ways i've set also fax detect for inbound and outbound on zapata, but that stills diferent from the receiving model as it relies on NVfaxdetect to detect. After many trials, i setup an architecture with another Server Running Hylafax and IAXmodem registring on my * Box and i just get out of troubles. It's perfect sending and receiving faxes with notifications and everything else, Hylafax + IAXmodem and Asterisk are working like a charm. I must say that we don't send too many faxes per day, but until now no problems! And yes didn't change anything on Zapata config or something else on Asterisk Box, i just added IAX account registred my Hylafax IAXmodem there and Voilá :) On 1/3/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Bill Gibbs wrote: My next step is to connect the fax machine to a Wildcard X100P. Check to see if there is Echo cancellation in the SPA-1001, and if so turn it off. If there is an adaptive Jitter buffer on the SPA-1001, try changing it to a fixed one (probably no more than 40ms). Why would you connect a fax machine to an X100P, aren't they FXO cards? Have you tried terminating to a VOIP provider? (to see if the problem is with the ATA). Here I use a fax machine connected to a CS6220 which is connected to the asterisk box and terminates with a TDM400P card (so a completely different arrangement). Any other suggestions? Black magic? Voodoo? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SNOM loses server registration
Hi Joao, I'm not very experienced with SNOM, but have you though about providing fix IP for you VoIP hardphones? That way you could avoid the registration problem. At least while you don't get your final solution. Hope it helps, MoutaPT On 1/3/07, Joao Pereira [EMAIL PROTECTED] wrote: Hello to all When my SNOM (300 or 320) loses Internet connectivity, it loses its Asterisk registration (ok, thats normal). But when the phone is back online, he doesn't try to register in Asterisk. I believe this happens to avoid flooding the private LANs when the Internet link is lost but the problem is that the phones don't try to re-register in the future Sometimes it stays 2 hours without registering to Asterisk. When this happens, the only solution is to reboot it (and hear the users complains) :( How can I avoid this? How can I reduce the time to re-register in SNOM 300 or 320 ? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Block some number outgoing from joust one extention
Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISA server Issue (Maybe off topic)
Hi! I have my Trixbox running behind a ISA server. However it works fine with Rix telecom (The service provider) The same setup dos not allow my phone trow the ISA server. It is seeing the phone as registering the public adress of the firewall instead of port forwarding it. anyone else had this issue? I am suing Sjphone and X-lite //Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: [asterisk-users] Disconnect supervision in India?
On 1/1/07, ram [EMAIL PROTECTED] wrote: On 12/30/06, Rajkumar S [EMAIL PROTECTED] wrote: On 12/29/06, Chris Earle [EMAIL PROTECTED] wrote: anyone know the status of disconnect supervision on POTS lines in India? Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have disconnect supervision.. It does not work afaik, you may not get caller id also. I tested upto 1.4b3 and no luck. its all depends on the provider where you take from. Does any provider's land line works well with TDM Cards? raj ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voice fax modem and asterisk
Hi I have been asked to ind out if there is a way to use asterisk to answere a voice fax modem so it can provide an answering service and record messages ? -- Gregory Machin [EMAIL PROTECTED] www.linuxpro.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] native music on hold distortion between files
I have native music on hold setup to play ulaw encoded files. No transcoding, caller is on a g.711u SIP channel. There is horrible distortion and noise between files for 1 to 2 seconds. Has anyone seen this? I check the files and trimmed silence from the end, the source of the noise is not the file. 1.2.13 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma Remora A202
Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. I did take the power cable from the CDROM to put on the card - I don't need the CDROM right now.. I'm looking for direction in getting this card working - I currently have a new Trixbox, hoping it'll have the software for this card already. If not, I'll be back asking what drivers I need. Sangoma seems to have a lack of documentation, but it may just be me thanks Todd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Hardware for Asterisk Server?
joe a. wrote: Mark Greene[EMAIL PROTECTED] Wrote on: 1/2/2007 12:58 PM: I believe I am going to start out with some refurbished Dell Poweredge servers. They have had a high success rate with a friend. I was going to go that route as well. But, depends on the model. I have several of the Poweredge 2300/2400 variety and these seem problematic. I could not get the final compile steps to perform on the 2400, for instance. Forget the exact issue. Also, these models, at least, do not directly support IDE drives, such as CD/DVD items. You are limited to SCSI versions, or trying to hack in an IDE controller. Which is fine, I guess, if all your source/install software is on CD. Or until the CDdrive fails and you have to hunt up a SCSI version. I've not seen, at any price, scsi versions of DVD drives. I am looking at the ACARD AEC7720-U IDE-SCSI bridge (converter) to get over that. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users lots of companies make scsi dvd drives -- g00gl3 is your friend... http://www.google.com/search?hl=enq=scsi+dvdbtnG=Google+Search signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Hardware for Asterisk Server?
Colin Anderson wrote: ASUS motherboards, in particular, have worked for me perfectly, everytime with both Digium and Sangoma cards. They are also easy to work with and well documented. -Original Message- From: Doug [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 02, 2007 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Hardware for Asterisk Server? At 07:20 1/2/2007, Mark Greene, wrote: Hey guys, In your experience what is the best way to go for a production asterisk box in your offices? With desktop prices so cheap you might think that you should just buy them off the shelf, but is that really a reliable machine? Anything you can tell me that would assist me in deciding the best way to obtain and maintain these boxes would be very helpful. I have even looked into building system myself that have no moving parts, but for about the same price I can build an immensely more powerful machine WITH moving parts. - Mark Case: 1 CodeGen 4U Server Case $80 http://tinyurl.com/bnobz http://tinyurl.com/95s2b Power Supply: 1 Dual 450 W. Power supply -- IStar https://www.ewiz.com/detail.php?name=PS-TC50R8A http://www.directron.com/tc400r8.html Motherboard, CPU 2GB of memory: http://www.mwave.com/mwave/skusearch.hmx?scriteria=MB-BA23083 http://www.mwave.com/mwave/viewspec.hmx?scriteria=BA21409 2 Hard Drives in RAID 1 config: http://www.mwave.com/mwave/viewspec.hmx?scriteria=AA48770 Digium card: 2 port, 64 bit, 3.3 volt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users no problems on my proliant DL580 signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Remora A202
It sounds like a bad card. Call Sangoma and ask them to replace it. You don't need to use the drive power cable for just a single fxo module. You only need it for the fxs or if you go over 2 fxo cards. In any case, it should not stop your computer from booting. Tom At 07:43 AM 1/3/2007, you wrote: Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. I did take the power cable from the CDROM to put on the card - I don't need the CDROM right now.. I'm looking for direction in getting this card working - I currently have a new Trixbox, hoping it'll have the software for this card already. If not, I'll be back asking what drivers I need. Sangoma seems to have a lack of documentation, but it may just be me thanks Todd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Remora A202
Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. Maybe your card is not properly seated. seems to have a lack of documentation, but it may just be me It is just you ;) http://wiki.sangoma.com/ If you still have problems with the card, contact Sangoma, they have very good customer support : http://www.sangoma.com/main/contact hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Remora A202
If the light on the dell is blinking amber... that typically means you have a power issue. Rob Time Bandit wrote: Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. Maybe your card is not properly seated. seems to have a lack of documentation, but it may just be me It is just you ;) http://wiki.sangoma.com/ If you still have problems with the card, contact Sangoma, they have very good customer support : http://www.sangoma.com/main/contact hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Hardware for Asterisk Server?
On Wed, 3 Jan 2007, Derek Whitten wrote: no problems on my proliant DL580 Nothing but problems with my DL380's until I ran a non-SMP kernel. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe() not recording calls
When I try to record a call the console shows: www*CLI Starting recording of MeetMe Conference 1 into file meetme-conf-rec-1-1167836078.0.wav. www*CLI The code being executed in extensions.conf is: exten = s,n(record),MeetMe(,rDMpc) ;Make new Room and record call. exten = s,n(bye),Playback(vm-goodbye) exten = s,n,Hangup The file never appears in /var/spool/asterisk/meetme Installed sw is: asterisk-1.2.14 kernel-2.6.18-1.2200.fc5.src.rpm asterisk-1.2.14.tar.gz lame-3.96.1 asterisk-addons-1.2.5lame-3.96.1.tar.gz asterisk-addons-1.2-current.tar.gz libpri-1.2.4 asterisk-core-sounds-en-gsm-1.4.3.tar.gz libpri-1.2-current.tar.gz asterisk-extra-sounds-en-gsm-current.tar.gz zaptel-1.2.12 asterisk-stat-v2_0_1.tar.gz zaptel-1.2.12.tar.gz Any ideas or thoughts on debugging would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma A102 w/ EC module gets intermittent echo /audio artifacts
I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: very bad phasing reverb feedback (from my rock roll days). This is quite intermittent, as in most cases, the user says, it's a one-time thing, they hang up, the problem caller calls back, everythings good. It's as if the Sangoma is trying too hard? I personally have not heard this, but I have to trust what the users say. Some ideas I'd like to bounce: 1. tx and rxgains - this card is plugged into an Atlas 550 which seems to run a little hot on the gains. 2. Timing - the card takes it's sync from the Atlas, which in turn syncs from the PRI. Maybe have one port on the card take it's timing from the other port? Don't see how it would be relevant, but hey, that's all I've got. 3. Taps? Does an A102 even care about taps or just echocancel=yes? Running * 1.0.9, Zaptel 1.0.9, FC2 yum-updated to current, quad Xeon. Production box, handles 2-6 thousand calls a day, snom 360 handsets w/ latest firmware, load average never goes over 2.0 My conf files: wanpipe1.conf: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 4 PCIBUS = 1 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 0 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES wanpipe2.conf: [devices] wanpipe2 = WAN_AFT_TE1, Comment [interfaces] w2g1 = wanpipe2, , TDM_VOICE, Comment [wanpipe2] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 4 PCIBUS = 1 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE = 2 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 2 TDMV_DCHAN = 0 [w2g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES Zaptel.conf: loadzone = us span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 Zapata.conf: [channels] language=en context=from-pstn switchtype=national pridialplan=unknown signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=yes txgain=-2.0 rxgain=-2.0 group=0 channel = 1-23 channel = 25-47 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fonebridge2
Hello List Does anybody have any experience with the FoneBridge line of products from RedFone? I think their HA implementation sounds interesting, and like the prospect of having dedicated hardware for our PRI connections. Kind Regards Jon Leren Schøpzinsky ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Hardware for Asterisk Server?
Steve Edwards wrote: On Wed, 3 Jan 2007, Derek Whitten wrote: no problems on my proliant DL580 Nothing but problems with my DL380's until I ran a non-SMP kernel. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users this one is 2x700mhz xeons 2gb ram running freebsd signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail to email
Hey guys, I need to set up asterisk so that it sends the voicemail to the users email. I understand that I need to say attatch=yes, but what else needs to be done. I would think that somewhere I need to specify the server that it uses to send the email, etc. - Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Hardware for Asterisk Server?
lots of companies make scsi dvd drives -- g00gl3 is your friend... http://www.google.com/search?hl=enq=scsi+dvdbtnG=Google+Search Well, who'd have thought? All my ususal suppliers said no one makes them. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo /audio artifacts ---More information
Aha, it just happened to me, so now I can characterize the audio: It basically sounds like it's missing every other sample - fuzzy and distorted. Timing? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [BULK] [asterisk-users] Fonebridge2
We tried them out early last year when we were looking at a large deployment and they gave us a lot of the redundancy that we wanted. However we did run into issues where calls seemed to get caught up in the system. It was as far as we could tell rather random. No consistency to it at all. Asterisk hung up the call but the telco side of the line didn't actually hang up. The channel was left open. Something was not being passed through the Fonebrige on to the telco to have the telco hang up the line. Sorry I'm not a guru on how phone systems work but the telco never received the proper hang up response from Asterisk. This caused channels to fill up fast. I confirmed this talking with our local company. They showed all 24 lines of our T1 in use yet Asterisk showed only 2 active calls. I talked with the gentlemen from Fonebridge and he was definitely helpful and more then willing to work with us to sort out the problem but sadly we just didn't have the time to wait to figure it out and went back to Digium cards. As soon as we put in the Digium cards the problem went away. You might want to check with them to see if they figured that one out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Schøpzinsky Sent: Wednesday, January 03, 2007 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [BULK] [asterisk-users] Fonebridge2 Importance: Low Hello List Does anybody have any experience with the FoneBridge line of products from RedFone? I think their HA implementation sounds interesting, and like the prospect of having dedicated hardware for our PRI connections. Kind Regards Jon Leren Schøpzinsky ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts
Zaptel.conf: loadzone = us span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 Just a quick thought in looking at the settings above, it appears that you have set both spans as the primary timing source. I am pretty sure that only one span should be the primary timing source. The other span should either be at 0 (not used as a timing source) or set as a secondary timing source. Hope this helps. Michael L. Young ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Remora A202
Try switching the order of the blank module and the FXO or remove the blank, I had a similar Dell do the same and after some experimenting found that the removing the blank solved the problem. On 1/3/07, Rob Schall [EMAIL PROTECTED] wrote: If the light on the dell is blinking amber... that typically means you have a power issue. Rob Time Bandit wrote: Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. Maybe your card is not properly seated. seems to have a lack of documentation, but it may just be me It is just you ;) http://wiki.sangoma.com/ If you still have problems with the card, contact Sangoma, they have very good customer support : http://www.sangoma.com/main/contact hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block some number outgoing from joust one extention
Hi Mattias, add this to your dialplan: exten= _/CALLERIDNUMBER,1,Hangup() ; Basically you are doing a pattern match with callerid match on your first priority! ; You may keep your remaining dialplan, no changes needed Pls Give me some feedback Best Regards, MoutaPT On 1/3/07, Mattias Andersson [EMAIL PROTECTED] wrote: Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail to email
There should be an example in your voicemail.conf Here is mine... mail is tagged from [EMAIL PROTECTED] and sent to [EMAIL PROTECTED] In voicemail.conf mailcmd=/usr/sbin/sendmail -f [EMAIL PROTECTED] [EMAIL PROTECTED] You of course would use the mailer that your system uses. I have sendmail on the same system as Asterisk. There are many other things you can define for mail but all should be in your example voicemail.conf Doug On Wed, 3 Jan 2007, Mark Greene wrote: Hey guys, I need to set up asterisk so that it sends the voicemail to the users email. I understand that I need to say attatch=yes, but what else needs to be done. I would think that somewhere I need to specify the server that it uses to send the email, etc. - Mark Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts
I think you are absolutely right. The audio I heard earlier sounds exactly like a timing issue. So: wanpipe1.conf: TE_CLOCK= NORMAL TE_REF_CLOCK= 0 wanpipe2.conf: TE_CLOCK= MASTER TE_REF_CLOCK= 1 zaptel.conf: span=1,1,0,esf,b8zs span=2,0,0,esf,b8zs I'm going to make this change and reload at lunchtime, I'll document it and post it to the list if it works. thanks for the good eye. -Original Message- From: Michael L. Young [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 03, 2007 9:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts Zaptel.conf: loadzone = us span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 Just a quick thought in looking at the settings above, it appears that you have set both spans as the primary timing source. I am pretty sure that only one span should be the primary timing source. The other span should either be at 0 (not used as a timing source) or set as a secondary timing source. Hope this helps. Michael L. Young ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] answer machine detection
Is there anyone with any experience of using the AMD app and the settings that worked for them in the UK ? Any help would be appreciated. Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo /audio artifacts
Colin Anderson wrote: I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: very bad phasing reverb feedback (from my rock roll days). This is quite intermittent, as in most cases, the user says, it's a one-time thing, they hang up, the problem caller calls back, everythings good. It's as if the Sangoma is trying too hard? I personally have not heard this, but I have to trust what the users say. Some ideas I'd like to bounce: 1. tx and rxgains - this card is plugged into an Atlas 550 which seems to run a little hot on the gains. 2. Timing - the card takes it's sync from the Atlas, which in turn syncs from the PRI. Maybe have one port on the card take it's timing from the other port? Don't see how it would be relevant, but hey, that's all I've got. 3. Taps? Does an A102 even care about taps or just echocancel=yes? I have fixed similar problems by reducing the gains. I call it ECFO Echo Cancel Freak Out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Dial out timeout
Hi, I am having a problem that is a miracle to me: If I dial out via voipstunt.com the call rings for a few seconds and then gives me a busy sign. - I do not have a timeout set in my dial command - the remote station does not cause the busy either - dialing the number with the voipstunt client does not give busy after a few seconds - dialing with the same voipstunt account with a softphone works without problems - when dialing out via other channels, i.e. iax or misdn on the asterisk machine, no timeout problem -in the CLI there is no message at all when the timeout occurs. It shows: ...snipp... -- Called [EMAIL PROTECTED] -- SIP/voipstunt-081b61a8 is making progress passing it to mISDN/1-1 P[ 1] After SETUP BC funke*CLI Given these facts I believe that the problem has something to do with my asterisk setup, and more specifically, as it only occurs with SIP, sip.conf. Is that reasonable? Unfortunately I have absolutely no idea, how to narrow it down further. My sip.conf looks as follows: -- [general] bindport=5060 bindaddr=0.0.0.0 srvlookup=yes allowguest=no qualify=no srvlookup=yes canreinvite=yes [voipstunt] type=friend host=sip.voipstunt.com disallow=all allow=g726 username=my_account fromuser=my_account secret=my_password qualify=2000 canreinvite=no promiscredir=yes rtptimeout=300 rtpholdtimeout=300 -- If anybody has any hint on how this might be solved, please let me know. Cheers. Arik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Dial out timeout
Arik Raffael Funke wrote: Hi, I am having a problem that is a miracle to me: If I dial out via voipstunt.com the call rings for a few seconds and then gives me a busy sign. Start out with not using the r option to the Dial line. That will remove the faked ringing tone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Power Specs
Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] API: how to bridge originated call?
(my pstn calls are coming in thru an upstream asterisk server, so the called and calling phone number is passed as an extension.) when caller comes in on 555, he will go to extension 1234 where he will wait for the API to make a call to 999 for him. how do I bridge the two calls? extensions.conf: ;context where caller comes in [caller] 555,s,1 Answer() 555,s,n UserEvent(Init) ;this lets me know the connection for 555 555,1234,1 Noop(caller waits to be bridged) 555,1234,2 Background(soothingmusic) ;context for connection - is this needed? [connect] from the API: (do I need to create a new context/extension first?) Action: Originate Channel: IAX2/upstream/999 -- calls 999222 thru upsteam IAX Context: ?? Exten: ?? Priority: ?? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power Specs
501 - 12V, 1A and a power/data cable 601 - 24V, 0.5A 650 - 24V, 0.5A - Dave On Jan 3, 2007, at 11:48 AM, Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power Specs
The 501 is 12VDC, and the 601 is 24VDC, as I recall. There was a post a few months ago that said that plugging the 24VDC into a IP501 will fry the phone. Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is chan_zap.so loaded?
Newbie question for sure... I'm unsure of how to tell if chan_zap.so is loaded. Also, does autoload in modules.conf take care of it or is it done explicitly? output of lsmod | grep zap: zaptel208388 16 wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,w ct4xxp,tor2 crc_ccitt 6465 1 zaptel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power Specs
FWIW, our Polycom IP601 phones use a transformer with output: 24VDC 500mA (center contact is positive). A Polycom reseller (or Polycom sales) could probably give you information on these other two models. Alvin Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is chan_zap.so loaded?
John French wrote: Newbie question for sure... I'm unsure of how to tell if chan_zap.so is loaded. Also, does autoload in modules.conf take care of it or is it done explicitly? output of lsmod | grep zap: zaptel208388 16 wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,w ct4xxp,tor2 crc_ccitt 6465 1 zaptel asterisk -rx show modules | grep -a zap ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Announce] Web-MeetMe 3.0.0 released
We've been holding back on this release to coincide with the Asterisk 1.4.0 release. This is mostly a compatibility release, but there are a few new features: * No longer requires register_globals in PHP * Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier) * Migrated all database interfaces to PEAR::DB which simplifies the code a bit and opens up the possibility of using other databases to host the scheduling DB (app_cbmysql is still only MySQL, but ODBC is planned/hoped for) * The conference monitoring code now uses the concise output from meetme list, improving the parsing of participant details. * Minor tweaks to improve the cbEnd.php script that enforces the conference duration, plays announcements and populates the conferencing CDRs. * Conference CDR records now store participant duration in seconds instead of a formatted string, allowing for further analysis (the web interface still formats the duration for display purposes) * App_cbmysql is updated to work with Asterisk 1.4.0 * App_cbmysql has it's own build environment now, no longer requiring a Makefile patch, etc... The new release can be found at: http://sourceforge.net/projects/web-meetme/ We do have a volunteer developer who will be maintaining the 2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and features that are not Asterisk version dependant will still be made available for older installations. Thanks, The Web-MeetMe development team... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Park and Page
I have a strange issue going on with one system. If you park a call and then do a page command, the parked call gets dropped. Both park and page use meetme, and it appears that the page uses the same conference number as parked call. So when the page is complete and hangs up, it drops the parked call. This is using Asterisk 1.2.13, Zaptel 1.2.11 and libpri-1.2.4. The system has a Sangoma A101 with wanpipe 2.3.4-2. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is chan_zap.so loaded?
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] John French wrote: Newbie question for sure... I'm unsure of how to tell if chan_zap.so is loaded. Also, does autoload in modules.conf take care of it or is it done explicitly? output of lsmod | grep zap: zaptel208388 16 wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,w ct4xxp,tor2 crc_ccitt 6465 1 zaptel asterisk -rx show modules | grep -a zap That's right - chan_zap is not a kernal module (as the .so name suggested). As such you won't see it with lsmod. You must use an Asterisk command. An alternative to grep is to use Asterisk's powerful command, e.g., asterisk -rx show modules like zap asterisk -rx show modules like chan_ or simply, asterisk -rx show modules like chan_zap Generally modules.conf should work. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Remora A202
Just a thought, maybe it won't boot because there is no power to the CD ROM drive [EMAIL PROTECTED] 01/03/07 8:43 AM Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. I did take the power cable from the CDROM to put on the card - I don't need the CDROM right now.. I'm looking for direction in getting this card working - I currently have a new Trixbox, hoping it'll have the software for this card already. If not, I'll be back asking what drivers I need. Sangoma seems to have a lack of documentation, but it may just be me thanks Todd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] API: how to bridge originated call?
I have uploaded a working patch for version 1.2.12.1, and other that seems to work in Trunk, but few people is reporting results, you can help to get this into Asterisk, go here: http://bugs.digium.com/view.php?id=5841 The patch I ported to 1.2.12.1 is working fine, I have tested in my servers, is the one called bridge-1.2.12.1.patch, there are other ones that say trunk, obviously only work with the trunk version of Asterisk. Kind Regards On 1/3/07, chester c young [EMAIL PROTECTED] wrote: (my pstn calls are coming in thru an upstream asterisk server, so the called and calling phone number is passed as an extension.) when caller comes in on 555, he will go to extension 1234 where he will wait for the API to make a call to 999 for him. how do I bridge the two calls? extensions.conf: ;context where caller comes in [caller] 555,s,1 Answer() 555,s,n UserEvent(Init) ;this lets me know the connection for 555 555,1234,1 Noop(caller waits to be bridged) 555,1234,2 Background(soothingmusic) ;context for connection - is this needed? [connect] from the API: (do I need to create a new context/extension first?) Action: Originate Channel: IAX2/upstream/999 -- calls 999222 thru upsteam IAX Context: ?? Exten: ?? Priority: ?? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 79x1 Auto-Answer
I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970 phones in a paging group. I have all the phones set up with an extra line that auto answers the dial from my paging extension when the primary line is not in use. All of these are operating correctly however the 7961/7970s all ring once and then auto answer so the person paging all the phones has the first part of his dictation clipped. This only happens with 7961/7970. The linksys and the older 7960 (running 7.4 sip firmware) all answer right away and they hear everything. Anyone have the newer 7961s and 7970s running 8.X SIP firmware auto-answering without any initial ring? Here are some configuration snippets: extensions.conf [globals] INTERCOM=Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED] [macro-page] ; Paging macro: ; Check to see if SIP device is in use and DO NOT PAGE if they are ; ${ARG1} - Device to page ; ${ARG2} - Other line (not paging line...we don't want to disturb their other line) ; exten = s,1,ChanIsAvail(${ARG2}|js) ; j is for dump and s is for ANY call to indicate busy exten = s,n,NoOp(${AVAILSTATUS}) exten = s,n,SIPAddHeader(Alert-Info: Bellcore-dr5) ; This is the shortest ringer for the Cisco phones exten = s,n,SIPAddHeader(Call-Info:\;answer-after=0) ; For Linksys SPA exten = s,n,NoOp() ; Add others here exten = s,n,Dial(${ARG1}||A()) exten = s,n,Hangup exten = s,102,Hangup() [page] ; Paging context exten = 2201_com,1,Macro(page,SIP/2201_com,SIP/2201) ; 7970 exten = 2202_com,1,Macro(page,SIP/2202_com,SIP/2202) ; 7961 exten = 2203_com,1,Macro(page,SIP/2203_com,SIP/2203) ; 7960 exten = 2204_com,1,Macro(page,SIP/2204_com,SIP/2204) ; 7960 exten = 2205_com,1,Macro(page,SIP/2205_com,SIP/2205) ; 7960 exten = 2207,1,Macro(page,SIP/2207,SIP/2207) ; Linksys SPA-942 Here are the configuration lines relevant to auto-answer in the 79x1/7970 configuration files: autoAnswerTimer0/autoAnswerTimer autoAnswerAltBehaviorfalse/autoAnswerAltBehavior autoAnswerOverridetrue/autoAnswerOverride This is configured under sipLines line in the XML file autoAnswer autoAnswerEnabled3/autoAnswerEnabled /autoAnswer The above phone configuration will allow the phone to auto-answer but after one ring. I'd like it to just work immediately. Any help would be appreciated. Jeremiah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error on answer a SIP 401 message
Hi, I'm a voip service provider and i'm setting up a asterisk box to register around 100 lines from my central softswitch. This asterisk box will be placed inside a customer and has a digium card to be interconected with customer's pabx. My problem is that when asterisk send register message, my softswitch return with sip 401 and asterisk should send a register message with Authorization in header. Only after 10, or 15 or 30 or 43 or 2 messages 401 asterisk start to send Authorization in header. This is a random time, don't follow any rule. This problem cause lines disregistration some times during a day. How can i solve this problem ? I use this parameters to register an account: register=number:[EMAIL PROTECTED]/number [fonar-number] type=friend context=default secret=pass username=number host=sip.provider.com fromuser=number fromdomain=sip.provider.com ;nat=yes ;insecure=very canreinvite=no ;qualify=1 dtmfmode=rfc2833 Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] API: how to bridge originated call?
By the way, Chester, please report results to the bug I sent you, is very imortant the users feedback to get this into Asterisk Regards On 1/3/07, Moises Silva [EMAIL PROTECTED] wrote: I have uploaded a working patch for version 1.2.12.1, and other that seems to work in Trunk, but few people is reporting results, you can help to get this into Asterisk, go here: http://bugs.digium.com/view.php?id=5841 The patch I ported to 1.2.12.1 is working fine, I have tested in my servers, is the one called bridge-1.2.12.1.patch, there are other ones that say trunk, obviously only work with the trunk version of Asterisk. Kind Regards On 1/3/07, chester c young [EMAIL PROTECTED] wrote: (my pstn calls are coming in thru an upstream asterisk server, so the called and calling phone number is passed as an extension.) when caller comes in on 555, he will go to extension 1234 where he will wait for the API to make a call to 999 for him. how do I bridge the two calls? extensions.conf: ;context where caller comes in [caller] 555,s,1 Answer() 555,s,n UserEvent(Init) ;this lets me know the connection for 555 555,1234,1 Noop(caller waits to be bridged) 555,1234,2 Background(soothingmusic) ;context for connection - is this needed? [connect] from the API: (do I need to create a new context/extension first?) Action: Originate Channel: IAX2/upstream/999 -- calls 999222 thru upsteam IAX Context: ?? Exten: ?? Priority: ?? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fonebridge2
Yeah, I've played with both an older one (FoneBridge 1) and a FoneBridge 2 unit, and they seem to work as advertised. The FoneBridge 2 is much nicer, sets up much faster and boots much faster. Besides, RedFone is great company to work with. Bill Burdick Hello List Does anybody have any experience with the FoneBridge line of products from RedFone? I think their HA implementation sounds interesting, and like the prospect of having dedicated hardware for our PRI connections. Kind Regards Jon Leren Schøpzinsky ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] have a phone number from stanaphone and a working trixbox, how do I connect them?
I have a phone number for traditional phone lines through stana phone and a working trixbox server. What do I need to do to connect the two so when someone calls the number from a normal phone, they get my server? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] over 200 queues, anyone?
Hello list, one of our clients is going to be deploying a system with over 200 differently composed queues and 100 agents. We are going to do a full test of the viability of this solution before deployment, but I was wondering if anyone has experience of such a setup and if there are any obvious problems or no-nos. Any suggestion welcomed, l. -- Home of QueueMetrics - http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] have a phone number from stanaphone and a working trixbox, how do I connect them?
I used these directions to get Stanaphone working on my FreePBX box: http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+6#614Stanaphone Alex On 1/3/07, blackwater dev [EMAIL PROTECTED] wrote: I have a phone number for traditional phone lines through stana phone and a working trixbox server. What do I need to do to connect the two so when someone calls the number from a normal phone, they get my server? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over 200 queues, anyone?
Yeah, get a Business Process specialist to analyze the client's environment and develop a better solution. 200 queues with only 100 agents sounds pretty ludicrous to me! On Wed, 2007-01-03 at 14:22 -0600, lenz wrote: Hello list, one of our clients is going to be deploying a system with over 200 differently composed queues and 100 agents. We are going to do a full test of the viability of this solution before deployment, but I was wondering if anyone has experience of such a setup and if there are any obvious problems or no-nos. Any suggestion welcomed, l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] have a phone number from stanaphone and a workingtrixbox, h
From: blackwater dev [EMAIL PROTECTED] I have a phone number for traditional phone lines through stana phone and a working trixbox server. What do I need to do to connect the two so when someone calls the number from a normal phone, they get my server? Thanks! Get a cheap X100P or a clone card. I had a Motorola SM56 winMODEM that worked fine, then bought an X100P from Digitnetworks (not affiliated with Digium). Clone cards can be found on eBay. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power Specs
The IP600 is 12v!!! I fried a 600 when I used power adapter from 601. On 1/3/07, Alvin Austin [EMAIL PROTECTED] wrote: FWIW, our Polycom IP601 phones use a transformer with output: 24VDC 500mA (center contact is positive). A Polycom reseller (or Polycom sales) could probably give you information on these other two models. Alvin Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power Specs
Peter, I have 600's that are 12V 1.5A, + in the center. This differs from some of the other answers, maybe those differences are regional (although that would seem rather silly). HTH B Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
And probably wont be as Steve Underwood explained to me that he is now supporting openpbx and has stopped support for unicall on asterisk 1.4 Can anybody at digium confirm? Is unicall going to be left out of 1.4? |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Carlos Chavez |Sent: Tuesday, January 02, 2007 6:02 PM |To: Asterisk |Subject: [asterisk-users] OnHook Call Announcement... | | I have a customer that is asking for a feature called On Hook Call |Announcement. The way he explains it is that when someone is on another call you can |sort of break in into their conversation but only the local person hears you and not the |external caller. | | Basically he wants to use this function so he can call anyone in the company |even if they are already on a call (he is the big boss). I saw that there is a feature |coming in 1.4 called Whisper paging that may do something like this but I need to know |if it is possible to do it in 1.2 because there is still no support for Unicall on 1.4 | |-- |Telecomunicaciones Abiertas de Mexico S.A. de C.V. |Carlos Chvez Prats |Director de Tecnologa |+52-55-91169161 ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power Specs
I haven't read trough the thread well enough. The 600 is 12V 1.5A indeed. Too bad they don't all have the same voltage. LST wrote: The IP600 is 12v!!! I fried a 600 when I used power adapter from 601. On 1/3/07, *Alvin Austin* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: FWIW, our Polycom IP601 phones use a transformer with output: 24VDC 500mA (center contact is positive). A Polycom reseller (or Polycom sales) could probably give you information on these other two models. Alvin Peder @ NetworkOblivion wrote: Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] yet another faxing issue (outbound only, via ATA)
I set aside a couple of channels and removed echo cancellation on them. So far, faxing outbound through an ATA is working fine now. Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Marco Mouta Sent: Wed 1/3/2007 6:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] yet another faxing issue (outbound only, via ATA) Hi all, I was having a similar issue, using TE110P from Digium all incoming faxes were detected and correctly received. When trying to send outbound faxes, they all get broken... I do believe it may be related with Echo Cancel enabled on my Zapata.conf, any ways i've set also fax detect for inbound and outbound on zapata, but that stills diferent from the receiving model as it relies on NVfaxdetect to detect. After many trials, i setup an architecture with another Server Running Hylafax and IAXmodem registring on my * Box and i just get out of troubles. It's perfect sending and receiving faxes with notifications and everything else, Hylafax + IAXmodem and Asterisk are working like a charm. I must say that we don't send too many faxes per day, but until now no problems! And yes didn't change anything on Zapata config or something else on Asterisk Box, i just added IAX account registred my Hylafax IAXmodem there and Voilá :) On 1/3/07, Thomas Kenyon [EMAIL PROTECTED] wrote: Bill Gibbs wrote: My next step is to connect the fax machine to a Wildcard X100P. Check to see if there is Echo cancellation in the SPA-1001, and if so turn it off. If there is an adaptive Jitter buffer on the SPA-1001, try changing it to a fixed one (probably no more than 40ms). Why would you connect a fax machine to an X100P, aren't they FXO cards? Have you tried terminating to a VOIP provider? (to see if the problem is with the ATA). Here I use a fax machine connected to a CS6220 which is connected to the asterisk box and terminates with a TDM400P card (so a completely different arrangement). Any other suggestions? Black magic? Voodoo? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over 200 queues, anyone?
Not necessarily... The same agents could very well be providing support for multiple companies. You wouldn't want an announcement from company A in company B's queues. Alex On 1/3/07, Joe Dennick [EMAIL PROTECTED] wrote: Yeah, get a Business Process specialist to analyze the client's environment and develop a better solution. 200 queues with only 100 agents sounds pretty ludicrous to me! On Wed, 2007-01-03 at 14:22 -0600, lenz wrote: Hello list, one of our clients is going to be deploying a system with over 200 differently composed queues and 100 agents. We are going to do a full test of the viability of this solution before deployment, but I was wondering if anyone has experience of such a setup and if there are any obvious problems or no-nos. Any suggestion welcomed, l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail to email
You just specify the users email address in the voicemail.conf file, along with their mailbox number: see the sample file: [default] ; Define maximum number of messages per folder for a particular context. ;maxmsg=50 2506 = 2506,Grandstream,[EMAIL PROTECTED],,attach=yes|imapuser=fbeck 1234 = 4242,Example Mailbox,[EMAIL PROTECTED] ;4200 = 9855,Mark Spencer,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=no|[EMAIL PROTECTED]|tz=central|maxmsg=10 ;4300 = 3456,Ben Rigas,[EMAIL PROTECTED] ;4310 = -5432,Sales,[EMAIL PROTECTED] ;4069 = 6522,Matt Brooks,[EMAIL PROTECTED],,|tz=central|attach=yes|saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes|envelope=yes|sayduration=yes|saydurationm=1 ;4073 = 1099,Bianca Paige,[EMAIL PROTECTED],,delete=1 ;4110 = 3443,Rob Flynn,[EMAIL PROTECTED] ;4235 = 1234,Jim Holmes,[EMAIL PROTECTED],,Tz=european 2503 = 2503,Forrest Beck,[EMAIL PROTECTED] On 1/3/07, Doug Crompton [EMAIL PROTECTED] wrote: There should be an example in your voicemail.conf Here is mine... mail is tagged from [EMAIL PROTECTED] and sent to [EMAIL PROTECTED] In voicemail.conf mailcmd=/usr/sbin/sendmail -f [EMAIL PROTECTED] [EMAIL PROTECTED] You of course would use the mailer that your system uses. I have sendmail on the same system as Asterisk. There are many other things you can define for mail but all should be in your example voicemail.conf Doug On Wed, 3 Jan 2007, Mark Greene wrote: Hey guys, I need to set up asterisk so that it sends the voicemail to the users email. I understand that I need to say attatch=yes, but what else needs to be done. I would think that somewhere I need to specify the server that it uses to send the email, etc. - Mark Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over 200 queues, anyone?
I stand corrected, but it still seems excessive. On Wed, 2007-01-03 at 15:06 -0600, Alex Robar wrote: Not necessarily... The same agents could very well be providing support for multiple companies. You wouldn't want an announcement from company A in company B's queues. Alex On 1/3/07, Joe Dennick [EMAIL PROTECTED] wrote: Yeah, get a Business Process specialist to analyze the client's environment and develop a better solution. 200 queues with only 100 agents sounds pretty ludicrous to me! On Wed, 2007-01-03 at 14:22 -0600, lenz wrote: Hello list, one of our clients is going to be deploying a system with over 200 differently composed queues and 100 agents. We are going to do a full test of the viability of this solution before deployment, but I was wondering if anyone has experience of such a setup and if there are any obvious problems or no-nos. Any suggestion welcomed, l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail to email
In my case it was never any confusion over what needs to be configured in asterisk. I was wondering what mail program asterisk used and what needed to be configured with it. In my case I had to set up sendmail on my system to relay through our internal mail server. sendmail.mc was the file I had to modify and SmartHost was what I had to append to. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ARI help
I am trying to use ARI for call monitoring. Recording conversations and such. The problem is that I don't use AMP, and don't have any sort of a database for CDR setup. It is all stored in the CSV file by default. When I setup ARI I tell it to go into standalone mode, and I set the asterisk manager username and password that was defined in manager.conf, but it also wants a cdr username and password that I don't know exists. Also, EVERYTIME I leave the callmonitor module active, it tells me that it could not find the DB extension and to check AMP, asterisk, and main.conf. So do I NEED to have AMP installed for call monitor to work? How can I setup ARI with JUST ARI and a STANDARD asterisk install. No AMP or SQL. - Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gentoo ebuild for 1.4?
Greetings list, Does anyone know if there's a maintained 1.4 ebuild for Gentoo? Even with the ~amd64 keyword, latest in the official Portage repository is 1.2.13. Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with centos 4.4 and jabber/gtalk (really iksemel)
Kenneth Padgett wrote: Bob, It looks like the gnutls development package is called gnutls-devel: 'yum install gnutls-devel' should get the package installed. Yah, I thought that would be it. I have that installed, as well as gnutls. (I basically installed both packages you can find with yum search gnutls). Any other thoughts, can I just d/l the libs and uncompress them somewhere? -Kenneth Kenneth, I don't have a Centos machine at home, but under Fedora Core 6 autogen.sh and ./configure work after installing gnutls-devel. I followed the instructions at: http://voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk as you suggested in your first post. Without gnutls-devel installed the autogen.sh step fails. I would have thought that FC6 and Centos 4.4 would be pretty close as far as directory hierarchy. Here is a list of the pertinent files from the FC6 gnutls-devel package: rpm -q --filesbypkg gnutls-devel gnutls-devel /usr/bin/libgnutls-config gnutls-devel /usr/bin/libgnutls-extra-config gnutls-devel /usr/include/gnutls gnutls-devel /usr/include/gnutls/compat.h gnutls-devel /usr/include/gnutls/extra.h gnutls-devel /usr/include/gnutls/gnutls.h gnutls-devel /usr/include/gnutls/openpgp.h gnutls-devel /usr/include/gnutls/openssl.h gnutls-devel /usr/include/gnutls/pkcs12.h gnutls-devel /usr/include/gnutls/x509.h gnutls-devel /usr/lib/libgnutls-extra.a gnutls-devel /usr/lib/libgnutls-extra.so gnutls-devel /usr/lib/libgnutls-openssl.a gnutls-devel /usr/lib/libgnutls-openssl.so gnutls-devel /usr/lib/libgnutls.a gnutls-devel /usr/lib/libgnutls.so You might want to check it against your Centos installation. If it's different try: ./configure --prefix=/usr --with-libgnutls-prefix=PFX Where PFX is the where libgnutls is installed (from ./configure --help). Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
Any screenshots available? I do not want to even test this without having any idea what it is or how it works. The brief description on sf.net is not enough. -- -- Steven http://www.glimasoutheast.org Dan Austin [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] We've been holding back on this release to coincide with the Asterisk 1.4.0 release. This is mostly a compatibility release, but there are a few new features: * No longer requires register_globals in PHP * Separated code from configuration settings in ./lib/defines.php (hopefully this will make future upgrades easier) * Migrated all database interfaces to PEAR::DB which simplifies the code a bit and opens up the possibility of using other databases to host the scheduling DB (app_cbmysql is still only MySQL, but ODBC is planned/hoped for) * The conference monitoring code now uses the concise output from meetme list, improving the parsing of participant details. * Minor tweaks to improve the cbEnd.php script that enforces the conference duration, plays announcements and populates the conferencing CDRs. * Conference CDR records now store participant duration in seconds instead of a formatted string, allowing for further analysis (the web interface still formats the duration for display purposes) * App_cbmysql is updated to work with Asterisk 1.4.0 * App_cbmysql has it's own build environment now, no longer requiring a Makefile patch, etc... The new release can be found at: http://sourceforge.net/projects/web-meetme/ We do have a volunteer developer who will be maintaining the 2.X.X chain for Asterisk 1.2.X compatibility, so bug fixes and features that are not Asterisk version dependant will still be made available for older installations. Thanks, The Web-MeetMe development team... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
On Wed, 2007-01-03 at 16:55 -0500, Steven wrote: Any screenshots available? I do not want to even test this without having any idea what it is or how it works. The brief description on sf.net is not enough. I'm testing the 2.0 version on asterisk 1.2 . What do you want to know about the application? Best regards, -- -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Remora A202
Thanks - that turned out to be the problem. Well- one of those solutions. I removed the blank and swapped the FXO module to the other port. I don't know if it was a bad port on the A200, but since I don't plan on using it, I won't worry about it- just regret it in a year when I get a second FXO module ;) As for documentation, I did find the info on WanPipe, but am not sure what Wanpipe is.. I'll do some more reading tonight. Thanks for the info. Todd On Jan 3, 2007, at 11:40 AM, Bruce Reeves wrote: Try switching the order of the blank module and the FXO or remove the blank, I had a similar Dell do the same and after some experimenting found that the removing the blank solved the problem. On 1/3/07, Rob Schall [EMAIL PROTECTED] wrote: If the light on the dell is blinking amber... that typically means you have a power issue. Rob Time Bandit wrote: Hi - I just got a Sangoma A200 card with a single 2FXO module and what appears to be an empty module. I put the card in my Dell GX260, but the power light on the front of the box just blinks and won't power up. Maybe your card is not properly seated. seems to have a lack of documentation, but it may just be me It is just you ;) http://wiki.sangoma.com/ If you still have problems with the card, contact Sangoma, they have very good customer support : http://www.sangoma.com/main/contact hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detect IP path before calling
Any easy way to determine if IP connectivity before attempting a SIP call? IP connectivity could be a timeout. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Core Dump in app_queue - Anyone seen?
Anyone seen this? It ocurred on a 'reload app_queue.so' command. Asterisk version is 1.2.9.1. Tried again, but it was not immediately reproducable. Doug. (gdb) bt #0 reload_queues () at app_queue.c:3339 #1 0xb778a7a8 in reload () at app_queue.c:4012 #2 0x0805bb44 in ast_module_reload (name=0x8137cc7 app_queue.so) at loader.c:257 #3 0x08092b3f in handle_reload (fd=33, argc=2, argv=0xbddfa470) at cli.c:147 #4 0x0809283e in ast_cli_command (fd=33, s=0x6d6f7250 Address 0x6d6f7250 out of bounds) at cli.c:1364 #5 0x080aef0f in action_command (s=0x81ead18, m=0xbddfaac0) at manager.c:927 #6 0x080b3ee4 in process_message (s=0x81ead18, m=0xbddfaac0) at manager.c:1305 #7 0x080b2ac5 in session_do (data=0x81ead18) at manager.c:1401 #8 0xb7f15ed8 in pthread_start_thread () from /lib/libpthread.so.0 #9 0xb7e147ea in clone () from /lib/libc.so.6 (gdb) bt full #0 reload_queues () at app_queue.c:3339 q = (struct ast_call_queue *) 0x81adca8 ql = (struct ast_call_queue *) 0xbddfaec0 qn = (struct ast_call_queue *) 0xb7dc03b3 cfg = (struct ast_config *) 0x81aca30 cat = 0x81507e0 mcao_QMain tmp = 0x6d6f7250 Address 0x6d6f7250 out of bounds var = (struct ast_variable *) 0x811e340 prev = (struct member *) 0x8101b79 cur = (struct member *) 0x2854554f newm = (struct member *) 0x0 new = 0 general_val = 0x2854554f Address 0x2854554f out of bounds interface = '\0' repeats 79 times penalty = 900 #1 0xb778a7a8 in reload () at app_queue.c:4012 No locals. #2 0x0805bb44 in ast_module_reload (name=0x8137cc7 app_queue.so) at loader.c:257 m = (struct module *) 0x81f3b10 reloaded = 2 oldversion = 863401873 reload = (int (*)(void)) 0xb778a7a0 reload #3 0x08092b3f in handle_reload (fd=33, argc=2, argv=0xbddfa470) at cli.c:147 x = 1 res = 1836020304 #4 0x0809283e in ast_cli_command (fd=33, s=0x6d6f7250 Address 0x6d6f7250 out of bounds) at cli.c:1364 argv = {0x8137cc0 reload, 0x8137cc7 app_queue.so, 0x0, 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0xbddfa49c h¥ß½ïÀÛ·h}\\bh}\\b, 0xb7dc3fea ë\234\211$ÿÐëÔ\213]ô\213uø\213}ü\211ì]é¿üÿÿU\211å\203ì(\211]ô\211uø\211}üè°Cûÿ\201ÃÜ\237\n, 0xb7e6fa00 , 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0xb7e6dff4 \034\020, 0x26 Address 0x26 out of bounds, 0x27 Address 0x27 out of bounds, 0xbddfa568 \200, 0xb7dbc0ef \213U\b\213\002\205Àu\b\213\205pÿÿÿ\211\002ÆD\aÿ, 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x26 Address 0x26 out of bounds, 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x0, 0x26 Address 0x26 out of bounds, 0xfbad8000 Address 0xfbad8000 out of bounds, 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227d8e , 0x8227dcc , 0x8227d68 ;\\b¬úæ·: 2007-01-03 15:17:39.165755\r\n, 0x8227dcc , 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0xbddfa544 ôßæ·\020\234 \bh¥ß½ê?Ü·, 0xb700 Address 0xb700 out of bounds, 0x0, 0xbddfa544 ôßæ·\020\234 \bh¥ß½ê?Ü·, 0xb7e6dff4 \034\020, 0x0, 0xb7e6da00 , 0x0, 0xb7f1a756 \201Ã\236H, 0xb7f1eff4 tî, 0xb7e6fa00 , 0xb7e6fa00 , 0xbddfa54c h¥ß½ê?Ü·, 0xb7f170eb ëÃ\213\203pÿÿÿ;(r\022\213\203Ðÿÿÿ;(s\b\213\203¤ÿÿÿë½\213\203 ÿÿÿ\213, 0xb7e6fa10 , 0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0xb7e6fa00 , 0xb7e6dff4 \034\020, 0xb7e6dff4 \034\020, 0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0xbddfa568 \200, 0xb7dc3fea ë\234\211$ÿÐëÔ\213]ô\213uø\213}ü\211ì]é¿üÿÿU\211å\203ì(\211]ô\211uø\211}üè°Cûÿ\201ÃÜ\237\n, 0xb7e6fa00 , 0x8209c10 È\017\025\bèÃ\035\b: 2007-01-03 15:17:39.165755\r\n, 0x21 Address 0x21 out of bounds, 0x21 Address 0x21 out of bounds, 0x81ead18 \017, 0x80 Address 0x80 out of bounds, 0x8091ffb \213\\$\030\203Ä\034ÃÇ\004$\004} e = (struct ast_cli_entry *) 0x81197a0 x = 2 dup = 0x8137cc0 reload tws = 0 #5 0x080aef0f in action_command (s=0x81ead18, m=0xbddfaac0) at manager.c:927 No locals. #6 0x080b3ee4 in process_message (s=0x81ead18, m=0xbddfaac0) at manager.c:1305 ret = 0 eqe = (struct eventqent *) 0x0 action = Command, '\0' repeats 72 times tmp = (struct manager_action *) 0x8144818 idText = ActionID: 2007-01-03 15:17:39.165755\r\n, '\0' repeats 217 times iabuf = 216.187.141.250 #7 0x080b2ac5 in session_do (data=0x81ead18) at manager.c:1401 m = {hdrcount = 3, headers = {Action: Command\000\n, '\0' repeats 238 times, Command: reload app_queue.so\000\n, '\0' repeats 225 times, ActionID: 2007-01-03 15:17:39.165755\000\n, '\0' repeats 217 times, \000\n, '\0' repeats 253 times, '\0' repeats 255 times repeats 76 times}}
RE: [asterisk-users] voice fax modem and asterisk
From: Gregory Machin [EMAIL PROTECTED] Hi I have been asked to ind out if there is a way to use asterisk to answere a voice fax modem so it can provide an answering service and record messages ? Absolutely - if that MODEM happens to be an X100P clone - such as my ENF656-PCIG-MOPR. There are quite a few - and you can buy one still. It can do a lot more than answering and recording. Yuan Liu -- Gregory Machin [EMAIL PROTECTED] www.linuxpro.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Remora A202
Thanks - that turned out to be the problem. Well- one of those solutions. I removed the blank and swapped the FXO module to the other port. I don't know if it was a bad port on the A200, but since I don't plan on using it, I won't worry about it- just regret it in a year when I get a second FXO module ;) No you won't, since Sangoma cards come with a 5 year warranty ;) Glad you fixed it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
Steven wrote: Any screenshots available? Sorry, not yet. I've been meaning to get the tools together to capture some images, but coding, QA and paid work have taken priority... I do not want to even test this without having any idea what it is or how it works. It is a suite of tools to permit the scheduling of conferences. The brief description on sf.net is not enough. There are four components to the suite: 1. A collection of web pages (PHP and javascript) that provide the interface to add, update, delete conferences. There is also a page to monitor active conferences, with the ability to mute, unmute and eject participants, as well as to have Asterisk place a call to a participant. 2. An Asterisk application that validates the conference id, start time, participant count and any user/moderator pins 3. A small php script that logs the conference participants to a CDR-like database and enforces a conference's scheduled endtime. 4. A small number of sound files to convey conference events/status. The files were recorded by Alison, so they are a nice match for Asterisk's standard sound files. Key features: 1. Enforcable start and end times. Optionaly alert callers if they are too early. Also optionally allow callers to join a conference if they are too early. 2. Configurable conference id, user and moderator pins. Pins are optional. If a moderator pin is set, then the user pin is required. 3. Recurring conferences. Dialy, weekly and bi-weekly 4. Future conferences can be edited. The entire series, or one at a time 5. A conference endtime can be extended once it has started, but this is the only configuration setting that can be changed once the conference start time has been reached. 6. Simple branding. Reasonably easy to change logos and page headings to refect your company (helps get past management objections) 7. Optional authentication (LDAP or Database) to permit users to schedule and manage their own conferences. I'll work on screenshots soon. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400 UK Caller ID problems ...
So just when I thought I had caller ID going fine things seem to have taken a turn for the worst. I'm now seeing lots of misses in picking up the caller ID on a line I know provides it. I know I've changed the TDM400 card and upgraded to the latest (1.2) version of Zaptel but could this be connected? This is what I'm seeing in the output: == Starting post polarity CID detection on channel 3 -- Starting simple switch on 'Zap/3-1' Jan 3 21:21:27 NOTICE[11667]: chan_zap.c:5888 ss_thread: Got event 17 (Polarity Reversal)... -- Executing NoOp(Zap/3-1, Look whos calling: ) in new stack (The NoOp line just dumps the caller ID, or not in this case) Sometimes it does work - which is what makes it more preplexing. Config files haven't changed and they have the right runes in them: usecallerid=yes cidsignalling=v23 cidstart=polarity One thing I have started doing is running fxotune on the lines - do you think this might make a difference? And has anyone any experience of CID on Telewest lines in the UK? (as opposed to BT which I'm currently using, but have a prospect who has some Telewest lines) Thanks, Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Any quiet 24 port POE switches out there?
I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect IP path before calling
With the chanisavail command. PaulH On Wed, 2007-01-03 at 14:22 -0800, Yuan LIU wrote: Any easy way to determine if IP connectivity before attempting a SIP call? IP connectivity could be a timeout. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over 200 queues, anyone?
lenz wrote: Hello list, one of our clients is going to be deploying a system with over 200 differently composed queues and 100 agents. We are going to do a full test of the viability of this solution before deployment, but I was wondering if anyone has experience of such a setup and if there are any obvious problems or no-nos. Any suggestion welcomed, one of our sites likes to micro manage things to the point of 38 queues. the thing you will find is that if your agents are members of various queues each member instance of each queue will get QueueMemberStatus events. (i've commented mine out because of flooding that occurs of the manager interface) deadlocks would be the thing to watch for. just food for thought. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any quiet 24 port POE switches out there?
On 1/3/07, John French [EMAIL PROTECTED] wrote: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The 8port Netgear switch on my desk doesn't have any fans. FS108p. Not sure if they make a 24port switch or not. -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...)
On 1/3/07, Anton Krall [EMAIL PROTECTED] wrote: And probably wont be as Steve Underwood explained to me that he is now supporting openpbx and has stopped support for unicall on asterisk 1.4 Can anybody at digium confirm? Is unicall going to be left out of 1.4? This has nothing to do with Digium, it has to do with anybody wanting to code the version for 1.4, AFAIK Steve never worked for Digium and Digium never distributed Unicall driver. Porting Unicall to 1.4 is in my TODO since 1 month ago, may be this month I will have the time to give a look at the code and try to make it work on 1.4, if somebody else cant do it before. Regards. -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any quiet 24 port POE switches out there?
I suspect any 24port will have a fan. The Netgear FSM7326P are not too bad and we have had good luck with them. ps - I also load their open source software. On Jan 3, 2007, at 4:51 PM, John French wrote: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Power Specs
Older models, 500 and 600, are 12V, newer 601s are 24v -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block some number outgoing from joust one extention
The easiest way is thru using contexts. On 1/3/07, Mattias Andersson [EMAIL PROTECTED] wrote: Hi all! I am shore someone have writing about it bout I cant find it. I have a extension that I need to block from making expansive mobil calls. Everyone else should be aloud to do the calls. I am shore it is possible to be done sens I had a commercial asterisk based PBX that I did that on. However I have switch to Trixbox because I need some custom functions not supported by the commercial product. I would appreciate all help. Regards Mattias Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] v140 ./configure not finding installed ssl
i'm building asterisk v140 on osx 10.4.8. openssl is installed in /usr/local/ssl, which openssl /usr/local/ssl/bin/openssl openssl version OpenSSL 0.9.8d 28 Sep 2006 asterisk is config'd with, % ./configure \ --prefix=/usr/local/asterisk \ --enable-shared \ --enable-static \ --with-ssl=/usr/local/ssl ./configure fails @, (...) checking for ssl2_connect in -lssl... yes checking /usr/local/ssl/include/openssl/ssl.h usability... no checking /usr/local/ssl/include/openssl/ssl.h presence... no checking for /usr/local/ssl/include/openssl/ssl.h... no configure: *** configure: *** It appears that you do not have the ssl development package installed. configure: *** Please install it to include OpenSSL support, or re-run configure configure: *** without explicitly specifying --with-ssl despite, checking, % ls -al /usr/local/ssl/include/openssl/ssl.h -rw-r--r-- 1 root wheel 79373 Sep 29 09:25 /usr/local/ssl/include/openssl/ssl.h fwiw, ssl is used widely/successfully elsewhere. suggestions as to what the issue is? thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Announce] Web-MeetMe 3.0.0 RE-released
While preping screenshots I found that a stupid little bug had slipped past my QA, relating to CDR views. I've fixed it and regenerated the tgz file and replaced the broken one on SF. If you have downloaded 3.0.0 today, please get a fresh copy. The bug was small enough and I think I caught it quick enough to avoid bumping the version up to 3.0.1 Thanks, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any quiet 24 port POE switches out there?
John French wrote: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. You will need a fanless switch like the 16 Port Netgear FS116P (8 port PoE and the rest are normal) http://www.tigerdirect.com/applications/searchtools/item-Details.asp?EdpNo=1697260sku=N100-2058 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error compiling chan_vpb
On 1/2/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: DiegoF wrote: chan_vpb.o:chan_vpb.cc:(.text+0x4da6): first defined here /usr/bin/ld: Warning: size of symbol `load_module' changed from 3274 in chan_vpb.o to 3926 in chan_vpb.oo collect2: ld devolvi el estado de salida 1 make[1]: *** [chan_vpb.so] Error 1 rm chan_vpb.o make: *** [channels] Error 2 hello, if somebody knows like solving this error, to him it will be been thankful. This has been fixed in Subversion branch-1.4; the fix will be included in the Asterisk 1.4.1 release. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you very much by the answer. And you know when you will leave this version?. And if he already left, in where I can find it?. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any quiet 24 port POE switches out there?
Am Mittwoch, den 03.01.2007, 16:51 -0600 schrieb John French: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. I had to browse through the list of switches on the market recently for different features. Most switches do not feature an acoustic entry in their description. Even those described as desktop devices... and just with it being named a desktop device does not necessarily give you a silent device, au contraire. All I found was the Nortel BAS220 48T (with 24 out of 48 ports PoE compliant), nominal 43.8 dB on the datasheet. I do not know that device, but noise information on PoE switches seems not to be a thing that manufacturers are proud of. I guess building a 1u-switch with an included 300W++ power adaptor requires active cooling, and the smaller the fans, the noisier the whirl. Maybe using several, smaller switches could do the trick for you. Brian Roy mentioned the Netgear FS108p (with external power adaptor, noiseless) as 8-port device. There is also a larger brother of it, the FS116P, which also comes with an external power supply, does PoE on eight of its 16 ports. I have no idea of your overall bandwidth requirements, but if it is only about phones, 100 MBit should be by far sufficient for those 20 devices, so you could cascade switches (like plugging two FS108p into non-PoE ports on a single FS116P, for instance). This is of course the cheapo way of doing it. Getting a proper multi-port switch, perhaps even a real brand one would be (ask the drooling sales droids out there) would be the real deal. rant Talking about NetGear switches, I once bought a 24port Gigabit Netgear switch, noiseless, external PSU. It was meant to be screwed to a table from below (in a classroom environment) with four metal brackets. The switch kept crashing (not letting any data through) in that environment, situation only changed when mounting that switch to a wall (with the CAT6 cables hanging straight down from the plugs) - temperature problem (which was not bad enough to go into warranty exchange. Just do not use the switch in a hot environment. 20°C in a boring computer lab) On a non-PoE device, with far less than 300W power to go through. /rant I personally do not trust wall-wart (a.k.a. external power supply) switches too much. I do not think it is a problem in principle, but those devices with internal power supply just tend to be better for me. YMMV. If you find something worthy, with a decent sound, please report back to the list so others can share a good experience. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 segfaulting when manager client is connected
I was just trying astman with the latest svn trunk from Dec 31. It connects, but if I attempt to make a call, asterisk segfaults, but in pthread_kill in /lib/tls/libpthread.so not in the asterisk code. Is this something others have seen? This is with glibc-2.3.4-2 I just upgraded to 2.3.6 (the lastest for Fedora core 3) and it's the same. Not much of a traceback, it's happening here: static struct eventqent *unref_event(struct eventqent *e) { struct eventqent *ret = AST_LIST_NEXT(e, eq_next); if (ast_atomic_dec_and_test(e-usecount) ret) pthread_kill(accept_thread_ptr, SIGURG); return ret; } Should I file a bug on this? I would presume if it's as trivial to duplicate as it is for me that others would have seen it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: [Announce] Web-MeetMe 3.0.0 released
I'm using the latest svn trunk code. The app_cbmysql builds, installs, and loads just fine. But I don't get any CBMysql application within asterisk. The only message I get as the module loads is something about finding the configuration file (successfully). The database tables are created, though the DDL scripts provided seem to require MySQL 5. The configuration file (cbmysql.conf) is in place and has been modified to reflect reality (username/pass and socket location). Any hints about what I must be doing wrong? Thanks, Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] API: how to bridge originated call?
Moises this sounds great! three questions if you don't mind: 1. how is this fitting into 1.6? 2. are there some directions I can follow for downloading the right source and applying your patches? 3. is there a central place for doc on your patches? (if not I would be more than happy to write it) thanks cy --- Moises Silva [EMAIL PROTECTED] wrote: By the way, Chester, please report results to the bug I sent you, is very imortant the users feedback to get this into Asterisk Regards On 1/3/07, Moises Silva [EMAIL PROTECTED] wrote: I have uploaded a working patch for version 1.2.12.1, and other that seems to work in Trunk, but few people is reporting results, you can help to get this into Asterisk, go here: http://bugs.digium.com/view.php?id=5841 The patch I ported to 1.2.12.1 is working fine, I have tested in my servers, is the one called bridge-1.2.12.1.patch, there are other ones that say trunk, obviously only work with the trunk version of Asterisk. Kind Regards On 1/3/07, chester c young [EMAIL PROTECTED] wrote: (my pstn calls are coming in thru an upstream asterisk server, so the called and calling phone number is passed as an extension.) when caller comes in on 555, he will go to extension 1234 where he will wait for the API to make a call to 999 for him. how do I bridge the two calls? extensions.conf: ;context where caller comes in [caller] 555,s,1 Answer() 555,s,n UserEvent(Init) ;this lets me know the connection for 555 555,1234,1 Noop(caller waits to be bridged) 555,1234,2 Background(soothingmusic) ;context for connection - is this needed? [connect] from the API: (do I need to create a new context/extension first?) Action: Originate Channel: IAX2/upstream/999 -- calls 999222 thru upsteam IAX Context: ?? Exten: ?? Priority: ?? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztdummy on 1.6
does anyone know if ztdummy is requires under 1.6 or are they using Linux' rtc? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [Announce] Web-MeetMe 3.0.0 released
Rob wrote: I'm using the latest svn trunk code. The app_cbmysql builds, installs, and loads just fine. But I don't get any CBMysql application within asterisk. The only message I get as the module loads is something about finding the configuration file (successfully). What version of Asterisk? What does this command return: *CLI cb mysql status The database tables are created, though the DDL scripts provided seem to require MySQL 5. The configuration file (cbmysql.conf) is in place and has been modified to reflect reality (username/pass and socket location). I would not expect the scripts to be MySQL 5 dependant, but they were contributed by another developer, so perhaps they are. How are you determining that you do not have the application? What Does this command report: *CLI show application CBMySQL Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 4
On Tue, Jan 02, 2007 at 03:17:35PM -0700, [EMAIL PROTECTED] wrote: Has anyone made this combination work together? I've tried everything and can't seem to get it work right. It all compiles fine, but when rxfax is called, I get an unknown symbol error. From my reading, everything points to me having multiple copies of spandsp and it's maybe calling the wrong one. After the complete compile of asterisk, I jump into the apps/ directory and do this: [~/asterisk/1.4/apps] [EMAIL PROTECTED]gcc -o app_rxfax.so -shared -Xlinker -x app_rxfax.o -lspandsp After that, with ldd on app_rxfax.so you can confirm that is is being linked. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any quiet 24 port POE switches out there?
Hi, I am using these model from HP ProCurve http://www.hp.com/rnd/products/switches/switch2600series/features.htm?jumpid=reg_R1002_USEN http://www.hp.com/rnd/products/switches/ProCurve_Switch_3500yl-5400zl_Series/features.htm?jumpid=reg_R1002_USEN Regards, Angel Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Mittwoch, den 03.01.2007, 16:51 -0600 schrieb John French: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. I had to browse through the list of switches on the market recently for different features. Most switches do not feature an acoustic entry in their description. Even those described as desktop devices... and just with it being named a desktop device does not necessarily give you a silent device, au contraire. All I found was the Nortel BAS220 48T (with 24 out of 48 ports PoE compliant), nominal 43.8 dB on the datasheet. I do not know that device, but noise information on PoE switches seems not to be a thing that manufacturers are proud of. I guess building a 1u-switch with an included 300W++ power adaptor requires active cooling, and the smaller the fans, the noisier the whirl. Maybe using several, smaller switches could do the trick for you. Brian Roy mentioned the Netgear FS108p (with external power adaptor, noiseless) as 8-port device. There is also a larger brother of it, the FS116P, which also comes with an external power supply, does PoE on eight of its 16 ports. I have no idea of your overall bandwidth requirements, but if it is only about phones, 100 MBit should be by far sufficient for those 20 devices, so you could cascade switches (like plugging two FS108p into non-PoE ports on a single FS116P, for instance). This is of course the cheapo way of doing it. Getting a proper multi-port switch, perhaps even a real brand one would be (ask the drooling sales droids out there) would be the real deal. Talking about NetGear switches, I once bought a 24port Gigabit Netgear switch, noiseless, external PSU. It was meant to be screwed to a table from below (in a classroom environment) with four metal brackets. The switch kept crashing (not letting any data through) in that environment, situation only changed when mounting that switch to a wall (with the CAT6 cables hanging straight down from the plugs) - temperature problem (which was not bad enough to go into warranty exchange. Just do not use the switch in a hot environment. 20°C in a boring computer lab) On a non-PoE device, with far less than 300W power to go through. I personally do not trust wall-wart (a.k.a. external power supply) switches too much. I do not think it is a problem in principle, but those devices with internal power supply just tend to be better for me. YMMV. If you find something worthy, with a decent sound, please report back to the list so others can share a good experience. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] API: how to bridge originated call?
1. how is this fitting into 1.6? 1.6? do you mean 1.4? AFAIK the most advanced Asterisk development goes in 1.4 2. are there some directions I can follow for downloading the right source and applying your patches? Nope, but is not hard at all. All the patches include the version in its name. But you need to learn how to use the patch command in Linux ( man patch ) and probably how to download code using SVN. 3. is there a central place for doc on your patches? (if not I would be more than happy to write it) Not really, since they are still under development and approval of Digium in bugtracker, how they work can change. But in the bugtracker there are some places where is explained how is supposed to work and be used. Check the link I sent you in my last email. Regards Moises -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] API: how to bridge originated call?
how is this fitting into 1.4? - can it be compiled against 1.4 or only 1.2? - if not, are there leanings in that direction? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users