Re: [Asterisk-Users] Asterisk with Grandstream ringback
- Original Message - From: Kim Lux [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 26, 2005 8:13 AM Subject: Re: [Asterisk-Users] Asterisk with Grandstream ringback I updated to firmware version x.22 and this and a few other problems were fixed. I was running x.18 and it allowed me to do a successful upgrade via http. Hi, could you please post your settings for http upgrade and url for firmware ? Thanks, Rob. On Tue, 2005-01-25 at 08:10 -0700, Kim Lux wrote: Are you saying that you are running firmware X.22 and it is not doing the callback when you hang up ? Where exactly did you get that firmware version ? Thanks On Tue, 2005-01-25 at 16:55 +0200, Doug Reid - Stormcorp wrote: Hi All Has any one tested Ver X.22 on the grandstreams? If so have you noticed the problem experienced with ringback? When you hang up the GS rings again and its the same call you put down. Only seen this with Ver X.16 and X.18 not yet with X.22 but I'm still not 100% convinced. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kim Lux, Diesel Research Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] size and quality of audio clipseffecttheplayback??
Ahh, here we are...got a little more detail: Linux ga1 2.4.21-27.ELsmp #1 SMP Wed Dec 1 21:53:57 EST 2004 x86_64 x86_64 x86_64 GNU/Linux Gabe - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 11:29 PM Subject: Re: [Asterisk-Users] size and quality of audio clipseffecttheplayback?? On Tue, 2005-01-25 at 23:14 -0800, Gabriel Afana wrote: Hi, This is what I am running: Red Hat Enterprise Linux ES release 3 (Taroon Update 4) Is the Taroon the kernel version? Do you think this could be a kernal issue (did you hear it for yourself at the site)? No Taroon I would have to guess is some internal name for the release. RH has always named their releases. No I didn't bother to listen to it. RH is broken with respect to decent kernels. Then again, I am a staunch debian supported and wouldn't ever use their kernel either. Try the new kernel and see. - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 4:02 PM Subject: Re: [Asterisk-Users] size and quality of audio clips effecttheplayback?? Have you compiled a vanilla kernel yet? I don't trust any distro supplied kernel. On Tue, 2005-01-25 at 12:07 -0800, Gabriel Afana wrote: Anybody have any ideas on this? I dont know what to do and my new website just launched yesteryday. www.gafana.com Go to the Real-time sport scores under the How It Works section. You can put your telephone number in there and it will call you. Listen to the message and hear what I am talking about. Its strange though, *just* right now I tried it and it sounded perfect...earlier this morning it jittered every few seconds! Gabe - Original Message - From: Gabriel Afana [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 24, 2005 4:21 PM Subject: [Asterisk-Users] size and quality of audio clips effect theplayback?? Hi, I've been having issues with asterisk playing back recorded messages. They sound clear..but there are lots of breaks during playback (like its losing packets). I got top-end hardware and I'm on a killer network so its not that. I've talked over my SIP line using a regular telephone and it sounds great, so its not the VOIP provider. Asterisk is working great with no other problems. So I'm thinking one of two things: More Obvious: The other thing I noticed is I get a warning on asterisk when I start the console saying the chan_oss it requested 8000 Hz but got 48000 Hz -- sound may be choppy. I am using the onboard sound card. Does Asterisk use a sound card to play the audio over VOIP or is sound card only needed if I have a physical phone hooked up to the computer? Less Obvious: Will the size and quality of a GSM audio file effect the playback? all my files were converted from wav files (22k 16bit stereo) to 8k mono GSMthe sound quality is fine, its just playback is choppy and wondering if playing with the actual GSM file format would change anything. Gabe -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BUSY-tone on incoming calls?
On Wed, 26 Jan 2005, Tobias Jönsson wrote: On Tue, 25 Jan 2005, Peter Svensson wrote: On Tue, 25 Jan 2005, Tobias Jönsson wrote: No, PRI_CAUSE works great at least in 1.0.2, probably in the earlier 1.0 releases too. Busy() may play a busy tone to the caller instead of signalling busy so using PRI_CAUSE is much better in PRI or BRI environment. The behaviour of Busy() and Congestion() can be changed with the priindication setting in zapata.conf. The options are inband (default) or outofband. This only affects the two applications mentioned above. Thank you for that information. I have now updated the wiki of zapata.conf. Still, like you said, it is better to explicitly set the PRI_CAUSE variable to the desired value. Isdn gives the user the ability to express problems etc in a detailed fashion. Might as wll use it. :) Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] size and quality of audio clipseffecttheplayback??
On Wed, 2005-01-26 at 00:07 -0800, Gabriel Afana wrote: Ahh, here we are...got a little more detail: Linux ga1 2.4.21-27.ELsmp #1 SMP Wed Dec 1 21:53:57 EST 2004 x86_64 x86_64 x86_64 GNU/Linux Dump that crap. Use a normal, vanilla kernel so you can avoid RH specific patches that are causing your problems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some more hardware and E1 questions
As long as the bootloader exists on both disks, and boot order are including both disks, there aren't any problems even booting with a failured disk. But since SATA is (often) Hot Plug, you could change the failed disk while running. - Original Message - From: Mark Eissler [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 7:50 PM Subject: Re: [Asterisk-Users] Some more hardware and E1 questions IMHO hardware RAID trumps software RAID. In order to use the latter your system must still be operational to some extent. -mark On Jan 24, 2005, at 11:10 PM, Gary wrote: better solution rather than have a machine with raid is to investigate ISCSI :-) On Mon, 24 Jan 2005 09:40:10 +0100, Daniel Nystrm wrote: I will be using Debian, and as long as the Linux Kernel supports the SATA controller, the rest shouldn't be any problems. If it's SATA RAID, I probably will use ordinary Linux software RAID, since it's more powerful than the simple one in the controller. - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 22, 2005 6:13 AM Subject: Re: [Asterisk-Users] Some more hardware and E1 questions Daniel Nystrm wrote: Hi again folks! ;) As before, I will transform one E1 30 Channel PRI into 30 FXS channels using Adit 600. Now I'm into choosing server platform. And the two opponents are: * Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1) * FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1) If you're planning to use SATA RAID on PE750, make sure your Linux distro supports. Your best bet - use Redhat Enterprise Linux or one of it derivatives. I'm using Centos 3, it autodetects the RAID whilst Mandrake 10 failed. As I've seen people having problem with HP server, I havn't looked at it at all. What experience do you have with the alternatives above? Which would you recommend? And another question at the same time; what's really E1? How is E1 devices connected? Seems like regular Cat5 cables, but it problably ian't? If anyone's using Adit 600, did they send all cables required for connecting to the FXS channels? Seems like a very unique plug on the side of Adit. Thanks! BR Daniel Nystrm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-biz] bellster.net - GREAT advance
Stewart == Stewart Nelson [EMAIL PROTECTED] writes: Stewart If this is not available, I would be willing to put some Stewart effort into enhancing the * MGCP stack, to also speak the Stewart slave side of the protocol. Are there other Free users that Stewart would be interested in contributing? Sure, count me in! Given that the Freebox hardware is the lowest-quality link in the chain (especially its echo cancellation), accessing Free's VoIP directly would be a big win and would free one FXO card in my PC. Tell me if you want me to setup a mailing-list for that. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Interesting bellster issue
dhh == dhickman [EMAIL PROTECTED] writes: dhh When I make a call, bellster anounces that I have no credits and dhh says goodbye, but it still routes the call. I just noticed another interesting problem: I checked that using Congestion I can appropriately reject an incoming bellster call and that another route is used (on extension +331, France, Paris). However, the second route tried by bellster ended up with This is 9:25 local time, calls are only permitted from ... to It means that the remote asterisk accepted the call to play the message, instead of using Congestion to use another route or fail. I lost one credit without having the call placed, but what is more important is that no other route has been tried, and that my PBX thinks that the call succeedeed and will not try an alternative route such a Zap line. The problematic route is 179. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] size and quality of audioclipseffecttheplayback??
- Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 26, 2005 12:14 AM Subject: Re: [Asterisk-Users] size and quality of audioclipseffecttheplayback?? On Wed, 2005-01-26 at 00:07 -0800, Gabriel Afana wrote: Ahh, here we are...got a little more detail: Linux ga1 2.4.21-27.ELsmp #1 SMP Wed Dec 1 21:53:57 EST 2004 x86_64 x86_64 x86_64 GNU/Linux Dump that crap. Use a normal, vanilla kernel so you can avoid RH specific patches that are causing your problems. Ok, I'll give it a try. Thanks for the info. Gabe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting a Wildcard TE110P working on E1's in Australia
Hello All, I have got my TE110P working at the hardware level, turned out to be a dodgy cable causing the Yellow errors in zttool. However, now I am getting yellow errors in asterisk, but zttool shows nothing out of the ordinary. here is my current config, and some asterisk console errors, any suggestions most welcome. zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 zapata.conf [channels] signalling=pri_cpe switchtype=euroisdn language=en context=sip channel = 1-15 channel = 17-31 echocancel= yes echocancelwhenbridged = yes echotraining = yes group = 1 usecallingpres= yes console errors: Jan 26 19:39:45 NOTICE[2158]: chan_zap.c:7381 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 Jan 26 19:39:45 WARNING[2158]: chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 1: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 2: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 3: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 4: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 5: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 6: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 7: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 8: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 9: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 10: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 11: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 12: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 13: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 14: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 15: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 17: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 18: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 19: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 20: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 21: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 22: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 23: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 24: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 25: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 26: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 27: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 28: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 29: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 30: Yellow Alarm Jan 26 19:39:45 WARNING[2159]: chan_zap.c:5642 handle_init_event: Detected alarm on channel 31: Yellow Alarm Jan 26 19:39:45 NOTICE[2158]: chan_zap.c:7381 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 Jan 26 19:39:45 WARNING[2158]: chan_zap.c:1925 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on channel 1 Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on channel 2 Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on channel 3 Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on channel 4 Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on channel 5 Jan 26 19:39:45 NOTICE[2159]: chan_zap.c:5637 handle_init_event: Alarm cleared on
Re: [Asterisk-Users] Re: Interesting bellster issue
Surely no other route would be tried in this instance, for as far as all devices are concerned the A party and B party were connected correctly, albeit in this instance to an announcement shelf device. I agree that the A party has a right to be annoyed at the loss of credit, but this has been tradition within telco's for as long as i can remember, as a call channel costs significantly more bandwidth than signaling The only time you don't lose credit (or get billed in traditional terms) is when the announcement shelf is contained within the same network as the A party. Why do you think that providers tend to offer free voicemail, to ensure every call is connected and further more get the call in the other direction It is however an interesting way of accruing free credits on the network. Food for thought David On 26 Jan 2005, at 08:30, Samuel Tardieu wrote: dhh == dhickman [EMAIL PROTECTED] writes: dhh When I make a call, bellster anounces that I have no credits and dhh says goodbye, but it still routes the call. I just noticed another interesting problem: I checked that using Congestion I can appropriately reject an incoming bellster call and that another route is used (on extension +331, France, Paris). However, the second route tried by bellster ended up with This is 9:25 local time, calls are only permitted from ... to It means that the remote asterisk accepted the call to play the message, instead of using Congestion to use another route or fail. I lost one credit without having the call placed, but what is more important is that no other route has been tried, and that my PBX thinks that the call succeedeed and will not try an alternative route such a Zap line. The problematic route is 179. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: cant do it in CLI anymore?
OK I found it in modules.conf. looks like this: ; Load either OSS or ALSA, not both ; By default, load OSS only (automatically) and do not load ALSA ; noload = chan_alsa.so ;noload = chan_oss.so Is this correct? cheers, Mick Jim Kou [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, Do you have load chan_oss/chan_alsa ? if not then you can't use 'Dial' app. Hope this help. :) Mick Hastings on 2005/1/26 03:31 wrote: Hi Floks, snip *CLI Dial No such command 'Dial' (type 'help' for help) *CLI the same thing for Answer, Hangup, etc what have I missed? cheers, Mick -- Jim Kou IT Engineer Malico Inc. Site: http://www.malico.com.tw No.5, Ming-Lung Road, Yang-Mei, Tao-Yuang, Taiwan 32643 Tel: +886-3-472-8155#2218Fax: +886-3-472-5979 __ ______ ___ _ _ _ ___ ( \/ ) /__\ ( ) (_ _)/ __)( _ ) (_ _)( \( )/ __) )( /(__)\ )(__ _)(_( (__ )(_)(_)(_ ) (( (__ (_/\/\_)(__)(__)()()\___)(_) ()(_)\_)\___)() ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with H323 channels
I trying to set up an h323 channel over TCP/IP network to connect two PBX. I just read http://www.voip-info.org/wiki-Asterisk+config+h323.conf but, it don't solve my dubs. How could I use a h323 channel with asterisk? Could anyone paste a part of h323.conf file? I am no sure how to setting up h323.conf. And the part of extensions.conf where you use the h323 channels for an specific prefix? Thanks.___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: cant do it in CLI anymore?
That's correct. Make sure that the chan_oss.so work properly, if so you can use Dial app. now. Mick Hastings on 2005/1/26 04:50 wrote: OK I found it in modules.conf. looks like this: ; Load either OSS or ALSA, not both ; By default, load OSS only (automatically) and do not load ALSA ; noload = chan_alsa.so ;noload = chan_oss.so Is this correct? cheers, Mick Jim Kou [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, Do you have load chan_oss/chan_alsa ? if not then you can't use 'Dial' app. Hope this help. :) Mick Hastings on 2005/1/26 03:31 wrote: Hi Floks, snip *CLI Dial No such command 'Dial' (type 'help' for help) *CLI the same thing for Answer, Hangup, etc what have I missed? cheers, Mick -- Jim Kou IT Engineer Malico Inc. Site: http://www.malico.com.tw No.5, Ming-Lung Road, Yang-Mei, Tao-Yuang, Taiwan 32643 Tel: +886-3-472-8155#2218Fax: +886-3-472-5979 __ ______ ___ _ _ _ ___ ( \/ ) /__\ ( ) (_ _)/ __)( _ ) (_ _)( \( )/ __) )( /(__)\ )(__ _)(_( (__ )(_)(_)(_ ) (( (__ (_/\/\_)(__)(__)()()\___)(_) ()(_)\_)\___)() ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Callmanager and Asterisk problem
Hello everybody, i got and asterisk and a CCM configured thru SIP, and in the sip show peers appear Name/usernameHostDyn Nat ACL Mask Port Status CCM 10.60.27.138255.255.255.255 5060 OK (1 ms) but when i enabled sip debug in the CLI got this Sip read: SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 10.60.0.136:5060;branch=z9hG4bK784b4a8c From: asterisk sip:[EMAIL PROTECTED];tag=as7b541ffe To: sip:10.60.27.138 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Content-Length: 0 7 headers, 0 lines Destroying call '[EMAIL PROTECTED]' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:10.60.27.138 SIP/2.0 Via: SIP/2.0/UDP 10.60.0.136:5060;branch=z9hG4bK4aaa1423 From: asterisk sip:[EMAIL PROTECTED];tag=as6f4153c7 To: sip:10.60.27.138 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 26 Jan 2005 09:15:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.60.27.138:5060 can anybody help me?, what could be the problem?? when i try to call an ccm extension got the busy signal, TIA Edgar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with H323 channels
I can recommend you to use the chan_oh323 from inAccess Networks - according to our experience it's much stable and bug free channel. http://www.inaccessnetworks.com/projects/asterisk-oh323 Lubo - AppRadius Project: Full RADIUS AAA support for Asterisk PBX http://appradius.minitelecom.org/ - [EMAIL PROTECTED] wrote: I trying to set up an h323 channel over TCP/IP network to connect two PBX. I just read http://www.voip-info.org/wiki-Asterisk+config+h323.conf but, it don't solve my dubs. How could I use a h323 channel with asterisk? Could anyone paste a part of h323.conf file? I am no sure how to setting up h323.conf. And the part of extensions.conf where you use the h323 channels for an specific prefix? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC Trunks
Hi all I have asked this question before but have not got any helping input. Im really new to this and need some explanation about ASTCC. So here is the question again. In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards. As I understand Brands is not used, Cards just makes the cards. Routed in the dialplan and pricelist, Trunks is for ASTCC to know where to terminate the call, the rest I dont know. What I can see in the trunks is that there only is IAX, SIP, Local and Zap available. What I need to use is H323 as I want to send all ASTCC calls to an H323 GateWay, How can I configure ASTCC/Asterisk to Make this possible? Any pointers and maybe config examples would be appreciated. Thanks KF ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom boot server problem
Hi, I'm trying to configure a Polycom IP Phone SoundPoint 500 to connect it to my Asterisk PBX but with no success. First of all, I downloaded the SoundPoint IP SIP Administration guide I found on internet and then I tried to make a boot server creating an FTP account on my Mandrake 9.1 Linux box but I needed the following files: .cfg sip.cfg phone1.cfg ipmid.cfg sip.ld so I searched inside polycom site (http://www.polycom.com/resource_center/1,,pw-492,00.html) and I found a link to the polycom resource center (http://extranet.polycom.com/csnprod/signon.html) but I hadn't a username and a password so I had to give up this way. I searched on internet and I found SoundPoint-IP_SIP_1.2.0.zip at this location: http://www.freedomphones.net/polycom/files/ Got the files I put them on my server ,I turned on the phone and then I set its boot parameters (server IP, username and password and a static IP for the phone) to point to my linux account. I connected the phone to my LAN (PC slot) and to my server test (which is not connected to the LAN, the idea is to use the other plug on the back side of the phone instead of a hub) through the LAN slot. No other choice is possible since the plugs are different. My config is: LAN --- SoundPoint 500(PC slot) SoundPoint 500(LAN slot) --- TestServer so I have connected all in this way: LAN --- SoundPoint 500 --- TestServer I ping-ed the phone from the test server and the phone answered. The first strange thing is neither the phone nor the test server can be ping-ed from another PC connected to the LAN, maybe some parameter are not correctly set in the phone config menu??? It is just like the phone isn't connected to LAN (but it is!). I went on this problem since I needed to make a test and the phone was seen by my test server. Pressing the about softkey during the boot countdown shows a lot of infos but the most important seems to be the last line: rev 2.0.2 30 Apr 02 16:33 Now, after restarting the phone, its screen shows: Welcome Initializing Phone ... Updating configuration but an error arises and I think it is correlated to 0004f2003cc2.cfg: the last line of this file is used by the phone to load the files it needs: APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone1.cfg, sip.cfg, ipmid.cfg MISC_FILES= LOG_FILE_DIRECTORY=/ The error showed on the phone screen is: Error saving application sip.ld while the log file on the boot server says: 0221043746|cfg |3|01|Updated bootrom configuration 0004f2003cc2.cfg. 0221043746|cfg |3|01|Updated file phone1.cfg. 0221043747|cfg |3|01|Updated file sip.cfg. 0221043750|cfg |3|01|Updated file ipmid.cfg. 0221043750|cfg |4|01|File is 4633471, which is bigger than file system.!! 0221043751|app1 |6|01|Error in saving application. but sip.ld HAS that sizemaybe I downloaded the wrong file??? So I tried to delete sip.ld from the server, maybe the phone didn't really need it but the phone complained: checking application Error saving application sip.ld while the log file says: 0221044153|cfg |4|01|Failed to load sip.ld. Check filename FTP parameters. 0221044153|cfg |5|01|Error updating app. 0221044154|app1 |6|01|Error in saving application. but FTP parameters are right because the log file is written on my boot server so the phone is really serching for sip.ld file. The last test I made was to delete sip.lf from the last 0004f2003cc2.cfg line: APPLICATION APP_FILE_PATH= CONFIG_FILES=phone1.cfg, sip.cfg, ipmid.cfg MISC_FILES= LOG_FILE_DIRECTORY=/ but the phone displayes: Error saving application while the log file on the boot server says: 0221050509|cfg |5|01|Error updating app. 0221050510|app1 |6|01|Error in saving application. After all these test I cannot go on. What is wrong with the phone setting? Is there something I forgot to do? Thanks in advance Giorgio __ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP600 stuck at Running App = sip.ld (was: Re: Polycom 1.4.1 firmware for IP500/IP600)
On Tue, Jan 25, 2005 at 10:00:42PM -0500, Robert Augustyn wrote: If you have it, can I get a copy please, or possibly can you send it to the keeper of http://www.freedomphones.net/polycom/files/ I am looking for the latest boot image too. 1) I have the 1.4.1 firmware. To whom should I send the files? There is no contact info in this web site. 2) Now I am having a problem with my IP600 test unit: While performing tests on the Polycom IP600 I changed a configuration item and during reboot the phone stopped at the Running App = sip.ld stage and seems stuck there. I reinitialized all configuration files to their defaults from the zip files you sent me, to no avail. Plugging/unplugging the phone does not help as it starts and then stops booting at the same stage, while the message waiting indicator stays solid red (whereas previously it would flash continuously until full startup). Bootrom version: 2.6.1 Sip.ld version: 1.4.1.0040 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-biz] bellster.net - GREAT advance
On Tue, 2005-01-25 at 15:55 -0800, Stewart Nelson wrote: If this is not available, I would be willing to put some effort into enhancing the * MGCP stack, to also speak the slave side of the protocol. Are there other Free users that would be interested in contributing? As another */Free user I'd be interested as otherwise I must put in other cards and put up with a long chain of conversions. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] interested in your opinion about FWD and iaxtel
Hello all, I am planning to connect my Asterisk with the FWD and/or iaxtel networks. Two mounths ago, I just used the iaxtel network, and i remember I have trouble with this network, I can not place a call. The service do not wotk. With FWD I alwais can place a call, I never get an error from this network. But my experience in both are very short, just a few test time. Could somebody experienced with any of this networks, told us what he/she thinks about? ¿witch one is better network?witch one has more users? Thanks for your time. Ismael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] optimumvoice
Hi. A friend of mine came asking about this VOIP provider. I havent heard of them so I thought I might ask the list. Anyone has any experience using them (http://www.optonline.net/Home, http://www.optimumvoice.com/index.jhtml)? Not just with Asterisk, even with their supplied HandyTone and a regular analog phone. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk to pstn
Sir/Mam, Good PM to all! I'm new to asterisk but I was able to setup a asterisk server using softphones. I have some questions in mind, I have a working asterisk server and I want to add digium cards w/ a telephone line. Will it be able to forward a call from the a person who is in the U.S. using a PC connected to a broadband dsl connection to my residence phone? If you don't want to receive spam, don't connect to the Internet, or don't have an e-mail address. Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue log analyser?
hi could I have a look at this? I really need it, urgently, so please.. roy On Jan 20, 2005, at 12:17, Ben Merrills wrote: I've not released the source yet, I asked last week on the mailing list for people to send me over some example queue_logs, because so far I've only been able to test the software against my own. I have however made a lot of changes to it since last I posted about it. Template engine has been improved Allows for recursion of a directory of templates Allows for different output directories (so you can do a daily, weekly and monthly all from the same set of templates say) And quite a few other bits As soon as I get some sample data that people don't mind the results being posted for then I can show it off a bit more. Hope to get some sample data soon, Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of João Amaro Sent: 20 January 2005 11:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] queue log analyser? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ben Merrills wrote: | There's a few (open source/free) ones in development. I myself am | developing one of them. | | Ben | Hi. Why not join all the project in just one ? Actually which queue log analyzers projects are beeing developed ? Check the mail from Ben Merrills sent to the list 14-10-2004 15:10. I don't know if he releases the source code, but, from the screenshots it seems to be a good one. Jo?o Amaro - -- Begin Mail | I've been doing some work on a queue log analyser for a while now, | getting the basics in place, an example of which you can find at | the URL below. However, just wondering what information people | think is most useful in a log analyser? | | At present it includes the following features: | | # Time periods - specify a period of days from the log which you | want to generate statistics for (e.g. only the last 14 days) # | Templating - allows the stats to be inserted into any html/text | template using specific tags to insert stats. This means you could | create a number of templates and execute the analyser against them | to give different information on different pages (quite flexible). | # Specify start and end dates - similar to the first feature, | except you can specify a tight period from your log, not just the | last x number of days # Channels/Agents to names - simple text file | allows you to specify a name, agent number and a channel - e.g. | Ben, Agent/1, Sip/ben. This is then used in the output # instead | of raw data # JPG graphs - includes a custom class to generate line | graphs of information (e.g. hourly call volumes etc) | | What I want to know though is, what output people would like. At | the moment there is an overview of all queues, which includes: | | Total Calls, total connected calls, total abandoned calls, calls | abandoned within x seconds, calls exited with key press, Average | hold time, max hold time, average talk time | | Agent overview includes: Calls taken, Average talk time | | Graph of call volume per hour of the day Graph of call volume per | day (over the period specified) | | Runs under windows (.NET or mono required) or any other OS that | support .NET/mono (Linux, Mac, BSD etc) | | http://muad.xdev.net/Projects/qig/sample.html | | | Not really done anything like this before, so as much input as | possible would be appreciated. | | Cheers, | | Ben -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFB75EfJUm/Bor63CERAv/UAKCwiYZ96RLqX0m7Ks9eL1f7iG4IDQCcCWvK gafg+vLAgQpjl75Hp5y8tug= =PwR8 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] setup questions- many users, little use
Hello All, Im on the technology committee for a fraternity at the University of Illinois. Were looking into moving from our current party line (one line shared between every two rooms) system to a PBX with voicemail in an effort to lower our monthly phone bill and provide better communication services. Weve pretty much settled on Asterisk as we do not wish to rewire all of our pots lines and cant justify $19,000 for Cisco Call Manager. We do not have many incoming/outgoing calls because most people are using their cell phones, but we do have to provide local service for 41 rooms, plus common areas and possibly remote users. Right now our setup is looking as follows: 12 ch T1 with 60 or 80 DIDs (using Digium T100P) P4 2.8/3 GHz Server (possibly a Dell 750, only running Asterisk) 41 Sipura ATAs (SPA-1001) 3 Cisco 7960s (in common areas, mostly to look good, but also to provide directory info) Our usage/requirements are as follows: Callers to main number will be greeted with IVR providing directory general announcements Members assigned one of DID numbers which will follow them their entire time living in, calls to DID numbers go strait through Voicemail for all users, as well as several general mailboxes Call groups based on committees (philanthropy, exec, alumni relations, etc) Call forwarding (only to internal extensions) Overhead paging/intercom Possible remote extensions for members living out of house (using 7960/40s) Management/configuration through our current portal system (web based, most likely well write this from scratch as we have pretty specific uses) Possible wake-up call scheduling Possible configuration from database already storing member information Future long distance service to Chicago area through collocation or similar I would greatly appreciate any input as to specific configurations, things to watch out for and consider, and any other useful information. Will our equipment selections work well with Asterisk? Will there be any compatibility issues with the T100P and the T1 from SBC? And what kind of reliability can we expect from this setup? Also if anyone has a setup similar to this please let me know how it worked out. We will most likely publish a case study, specific configuration guide, and extensive documentation after we finish implementation, as other fraternities and sororities on campus have also expressed interest in our approach to IT management. Thanks in advance for any help, Bill Lattner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [Fwd: Re: [Asterisk-biz] bellster.net - GREATadvance]
As far as I know it's not legal to join bellster in Israel. It means that you're reselling the minutes you buy from the telco company. It also means that you need a permit from the Israeli ministry of communications cause you're acting as an international call provider. Can't be done here. -Original Message- From: Geoffrey S. Mendelson [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 25, 2005 5:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [Fwd: Re: [Asterisk-biz] bellster.net - GREATadvance] On Tue, Jan 25, 2005 at 02:43:27PM +1100, Duane wrote: Another small point is that a lot of countries don't have flat rate calls, and I highly doubt anyone in those countries would be offering their land lines for this kind of service either. It costs me between 20 and 30c per call to make local calls, so this basically only leaves North American and New Zealand as the only viable options that I know of. The situation is the similar in Israel. No calls are cheap, calls to cell phones are 3-4 times the cost of calls to landlines. The local cable company is offering cheap VOIP (but not in Jerusalem yet), but I'd hate to see the combined latency. Geoff. -- Geoffrey S. Mendelson, Jerusalem, Israel [EMAIL PROTECTED] N3OWJ/4X1GM IL Voice: 972-544-608-069 IL Fax: 972-2-648-1443 U.S. Voice: 1-215-821-1838 I may be an old fart, but I'm a high-tech, up to date old fart. :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] optimumvoice
Being a CableVision customer I get harrassing phone calls from these guys all the time trying to sell me their OV service. Firstly it's closed. They won't allow you to bring anything to their network. Secondly it uses G729 so there's no faxing etc (although you can buy that for extra cost). Finally its $34.95 all in. Very expensive in light of others like Vonage and GalaxyVoice. They do have porting abilities and a 911 service which some others don't. Mark Shoval Tomer wrote: Hi. A friend of mine came asking about this VOIP provider. I havent heard of them so I thought I might ask the list. Anyone has any experience using them (http://www.optonline.net/Home, http://www.optimumvoice.com/index.jhtml)? Not just with Asterisk, even with their supplied HandyTone and a regular analog phone. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Damn DTMF Beeps on my calls
Well, it sounds to me like his phone actually IS sending you keypresses. You stated that it goes silent on his end while you are hearing his DTMF tones. Sounds like the phone is silencing his end, as it would if he were intentionally dialing out. I am guessing that he has a bad phone or something on his end is causing the issue. Sincerely, Texas Web Geek -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Me Sent: Wednesday, January 26, 2005 1:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Damn DTMF Beeps on my calls Well this happens a LOT when I call one particular person, not so much when I call others. Both sides of the call are running Sipura ATA's with * in the middle, no termination or Zap in between at all. It seems that when I call this person from my home address it occurs a LOT like 1 or 2 times a minute or more at times. When I call from another location, same ATA type but different building it doesn't happen (I don't think). The other caller does not here it at all, he only hears silence when I hear the beep. It sounds EXACTLY like a key being pressed on my phone. It's not just a beep, to answer the other posters question. -- Start Your Own ISP! http://www.YourOwnISP.com - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 24, 2005 2:38 PM Subject: Re: [Asterisk-Users] Damn DTMF Beeps on my calls On Mon, 2005-01-24 at 13:38 -0600, Me wrote: Can someone give me a clue as to why I keep hearing DTMF type beeps on my phone calls. It sounds exactly like someone on the other end is pushing a key on their phone but they are not! Has anyone ever heard of this before? It use to happen once in a while, today it's been happening a LOT and it's driving me batty.. As usual, if you want to ask a smart question you need to add more details. DTMF can be caused by talk off. Essentially a voice pattern that triggered the DTMF detection. Now for the part that would have been smart, identifying the location your DTMF is being detected. If it where all zap, then it is the DTMF routines in asterisk/zapata, but as you didn't bother to expound what is going on, it could be SIP hardware phones acting up on you. More details please before you go batty. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] interested in your opinion about FWD and iaxtel
Ismael Gil wrote: I am planning to connect my Asterisk with the FWD and/or iaxtel networks. There is nothing stopping you from connecting to both simutaniously, in general there's only small amount of overhead to remain connected to a foreign network. Although there is no overhead from listing yourself with e164.org. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers I do not try to dance better than anyone else. I only try to dance better than myself. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tie web application to VOIP
- Original Message - From: K J [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 24, 2004 10:06 PM Subject: [Asterisk-Users] Tie web application to VOIP I want to tie my web application (built using .NET + MS SQL Server) into a VOIP service so that users can call each other. I want them to interface with my application's username system. On the receiving user's end, he can either receive the call using a VOIP phone, or windows application (like skype). I would use Skype's API, but it appears I need to use their username system, and not my own. My question is, what software/hardware solutions would I need to do this? Any suggestions/feedback would be greatly appreciated. Btw, I was told that Asterisk + SER would do the trick. However, I'm a newbie to the world of VOIP. If someone can give me some tips/hints, it would be great. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Issue with res_config_mysql.so in latest CVS
Hello, I just checked out the latest CVS and compiled and now get the following error: [res_config_mysql.so] = (MySQL RealTime Configuration Driver) Jan 26 13:03:51 WARNING[27081]: config_old.c:27 ast_load: ast_load is deprecated, use ast_config_load instead! == Parsing '/etc/asterisk/res_mysql.conf': Found Jan 26 13:03:51 WARNING[27081]: res_config_mysql.c:561 parse_config: MySQL RealTime: No database socket found, using '/tmp/mysql.sock' as default. Jan 26 13:03:51 WARNING[27081]: config_old.c:39 ast_destroy: ast_destroy is deprecated, use ast_config_destroy instead! asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: ast_cust_config_register Any ideas on how to resolve this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP QoS with PIX
Hi List Just a little bit OT, but then again perhaps an information that could be of great value for a lot of administrators !! Does anyone have experience with how to setup VoIP QoS for outgoing data through a Cisco PIX (515) ? I believe that it should be possible to give higher priority to outgoing VoIP packets. This is due to the problem of ADSL as the UpStream data rate is 1/4 of the DownStream data rate. Regards BennyB ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS for Asterisk
Asterisk is software installed on linux installed in a PC with a hard drive. When I say it might not come up after a power failure I don't mean Asterisk, I mean Linux. The hard drive might fail and you can kiss you system good bye. Legacy PBXs don't have that problem. The configuration there is on NVRAM or Flash, and when the power comes back up it just loads and keeps working. A UPS for legacy PBX means that if the power outage is two minutes long, you can keep talking as if nothing happened. For Asterisk it's a must. -Original Message- From: David Brodbeck [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 25, 2005 9:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] UPS for Asterisk -Original Message- From: Shoval Tomer [mailto:[EMAIL PROTECTED] That's not a problem. The question is what happens when the power's restored. Can you go ahead and just start working or do you need to call the technicians to come reconfigure the whole thing? It comes back up on its own, of course. If it just works, you have something asterisk without UPS can't give you. Really? Surely Asterisk can be configured to start itself up when the system boots. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 - channel out to lunch?
Today I had channel 1 on my TDM400 go to sleep, verified by pluggin in known good telco lines in various combinations on channel 1 through 4 - problem is channel 1, not anything external. So after seeing lots of stuff on the list re: TDM400's I power cycled, removed board and let linux say nothing's home, replaced board, told linux to ignore it etc. No OS/asterisk/etc changes made on the box between the it worked stage and it stopped working stage. Any suggestions would be welcome - and it just so happens that a previously RMA'd TDM board should be here tomorrow - so when that one arrives I'll swap modules and see if it follows the module or the board... For those of us that have had probems with the tdm dropping, it seems stopping *, stop and restart zaptel, restart * fixes what seems to be a software bug. No reboot necessary. If that doesn't fix the problem, then you might have a defective module. There was an issue with the first tdm cards shipped (ver e/f) where the first module slot had a problem. Those that received replacement cards found an added jumper wire on them suggesting a printed circuit board trace had been missed (or something like that). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS for Asterisk
OK. You're wearing me out. IF linux boots Asterisk can surely load automatically. What if linux DOES NOT boot after a power failure? -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 25, 2005 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UPS for Asterisk David Brodbeck wrote: It comes back up on its own, of course. If it just works, you have something asterisk without UPS can't give you. Really? Surely Asterisk can be configured to start itself up when the system boots. Absolutely. Where did the notion ever come from that it could not? B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot get call transfers working
I have installed asterisk from the CVS source on Jan 7th and I am having
problems getting call transfers working.
features.conf contains:-
[featuremap]
blindxfer = #1 ; Blind transfer
disconnect = *0; Disconnect
automon = *1 ; One Touch Record
atxfer = *2; Attended transfer
extensions.conf contains :-
[macro-iax]
; Macro to define a default call to IAX2 connected extensions
exten = s,1,Dial(IAX2/${MACRO_EXTEN},20,tT)
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
We are using Firefly with the IAX2 protocol for each end user.
When I call one extension from another I am unable to transfer a call.
If I type *2 quickly nothing happens.
Any suggestions?
Thanks
Gareth
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RE: [Asterisk-Users] UPS for Asterisk
:) Get an APC power switch, hook it up to a network in the vicinity of your * box and make sure you can reach it from the outside. If the box fails to boot you can remotely power cycle it. If you need a rocksolid solution have a look at astlinux that can boot * from a compact flash card in read only mode which makes it very hard to break :) On Wed, 26 Jan 2005, Shoval Tomer wrote: OK. You're wearing me out. IF linux boots Asterisk can surely load automatically. What if linux DOES NOT boot after a power failure? -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 25, 2005 10:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UPS for Asterisk David Brodbeck wrote: It comes back up on its own, of course. If it just works, you have something asterisk without UPS can't give you. Really? Surely Asterisk can be configured to start itself up when the system boots. Absolutely. Where did the notion ever come from that it could not? B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with PSTN Help........needed!!!!!!!
hi all, Im trying to configure a * server with FXO and FXS cards. Basically what i want to do is be able to recieve calls from PSTN and dailout as well...but im really very confused with how to handle the 30 channels coming in on the PRIwhat would be the best hardware to use n stuff.. if someone can give me some hints or config examples that would be just great Thank's in advance... Umair Network Engineer Hello Technologies. 1-703-857-6230 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disa Syntax, some help please
Hi I'm confused by the asterisk WIKI syntax for DISA. I want to only let a CID of say 1234567 pass through DISA, which calls an extension of 333 In reading the documentation, I thought it should look like this exten = 333/1234567,1,Authenticate(1234567) exten = 333/1234567,2,DISA(no-password|my_context) This throws up all sorts of errors. I simply want the callerid to be tested, if its correct, the user should pass though DISA and onto my_context Any help here would be appreciated ! -- Open WebMail Project (http://openwebmail.org) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS for Asterisk
If you need a rocksolid solution have a look at astlinux that can boot * from a compact flash card in read only mode which makes it very hard to break :) You should be able to boot Asterisk using slackware as a base from a 64M CF card or even from a 64M bootable USB memory key. If you use ReiserFS or something similar for the drive that stores all your voicemail, etc then it should come back without a problem as well. Of course you want to make sure the system shuts down cleanly too... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Interesting bellster issue
David == David John Walsh [EMAIL PROTECTED] writes: David I agree that the A party has a right to be annoyed at the loss David of credit, but this has been tradition within telco's for as David long as i can remember, as a call channel costs significantly David more bandwidth than signaling I totally agree, but it would be more honest to reject the call instead of accepting it and play the message; I am not concerned by the cost but by the fact that no alternative route will be tried in this case, even if one would be available. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with PSTN Help........needed!!!!!!!
Hello, FXO and FXS cards are only for analogic lines, but if you need connect * with a PRI, maibe need añother kind of hardware. See the digium or AVM homepages. There you'll find what are you looking for. Ismael. On Wed, 2005-01-26 at 17:31 +0500, Hussain Umair wrote: hi all, Im trying to configure a * server with FXO and FXS cards. Basically what i want to do is be able to recieve calls from PSTN and dailout as well...but im really very confused with how to handle the 30 channels coming in on the PRIwhat would be the best hardware to use n stuff.. if someone can give me some hints or config examples that would be just great Thank's in advance... Umair Network Engineer Hello Technologies. 1-703-857-6230 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New ip billing solution?? any updates?
Well, in our country (dont know others) we have different plans for residential users, other plans for commercial users, about 8 international long distance phone services that you can select at dialing time and 3 carriers for domestic long distance.Ohh, and our cellphone providers (TDMA/CDMA/GSM) have different rates (guess this is a rate hell but im sure it got to be worst somewhere else) I am looking for something that lets me plan different providers according to route cost and how to configure * to do so, as well as to handle stuff like today we got 500 minutes at 0.01 but tomorrow at 0.007 so routes must know that today this provider is expensive compared to others but tomorrow it might not. Savinovich, do you have some PDF i can see, or demo? Thanks, On Tue, 25 Jan 2005 16:43:07 -0500, Paul Rodan [EMAIL PROTECTED] wrote: www.bicomsystems.com has a pretty nice billing system built into it, and it's Asterisk based. Not sure if they sell it standalone. We use a mom and pop cdr type of system. We modified cdr_mysql.c to separate national/international and incoming toll free calls into a separate mysql database. Then we use a perl script to read it in, as well as a rate table, do the math and inject the amount the customer owes us into our older billing system which sends out the bills. It can adjust for international calls placed to cell phones or regular city calls, match the international destination, etc. It adjusts for each customer by the account code. I didn't think it was too bad. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez Sent: Tuesday, January 25, 2005 3:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] New ip billing solution?? any updates? Hi people, i've seen the wiki looking for a * billing solution but the links point to websites that have not updated their content (or news) section for over a year. Can anyone recommend a commercial-grade (i mean no mompop cdr system) billing solution that can start small and then scalate as traffic grows and tested/used with Asterisk before? commercial or open source links are ok. thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP600 stuck at Running App = sip.ld(was: Re: Polycom 1.4.1 firmware for IP500/IP600)
Did you try to boot without lan just the power ... I've had this same problem to and rebooted the device without lan connection -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Louis-David Mitterrand Sent: woensdag 26 januari 2005 11:36 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Polycom IP600 stuck at Running App = sip.ld(was: Re: Polycom 1.4.1 firmware for IP500/IP600) On Tue, Jan 25, 2005 at 10:00:42PM -0500, Robert Augustyn wrote: If you have it, can I get a copy please, or possibly can you send it to the keeper of http://www.freedomphones.net/polycom/files/ I am looking for the latest boot image too. 1) I have the 1.4.1 firmware. To whom should I send the files? There is no contact info in this web site. 2) Now I am having a problem with my IP600 test unit: While performing tests on the Polycom IP600 I changed a configuration item and during reboot the phone stopped at the Running App = sip.ld stage and seems stuck there. I reinitialized all configuration files to their defaults from the zip files you sent me, to no avail. Plugging/unplugging the phone does not help as it starts and then stops booting at the same stage, while the message waiting indicator stays solid red (whereas previously it would flash continuously until full startup). Bootrom version: 2.6.1 Sip.ld version: 1.4.1.0040 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 stuck at Running App = sip.ld (was: Re: Polycom 1.4.1 firmware for IP500/IP600)
If you have it, can I get a copy please, or possibly can you send it to the keeper of http://www.freedomphones.net/polycom/files/ I am looking for the latest boot image too. 1) I have the 1.4.1 firmware. To whom should I send the files? There is no contact info in this web site. 2) Now I am having a problem with my IP600 test unit: While performing tests on the Polycom IP600 I changed a configuration item and during reboot the phone stopped at the Running App = sip.ld stage and seems stuck there. I reinitialized all configuration files to their defaults from the zip files you sent me, to no avail. Plugging/unplugging the phone does not help as it starts and then stops booting at the same stage, while the message waiting indicator stays solid red (whereas previously it would flash continuously until full startup). Bootrom version: 2.6.1 Sip.ld version: 1.4.1.0040 When I was testing some of the Polycom phones a month or two ago, I had problems loading software a well. I had initially tried using a tftp server (even though several notes strongly suggested using ftp only), and tftp didn't work. I moved to ftp, and ran into problems as well. The Polycom phones look at each file's timestamp, and if it is equal-to-or-older-than what is running on the phone, it won't load it. Don't know if that is the same problem that you're seeing, but you can easily test that by simply touching the file to give it a more recent timestamp. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Am i in control after i dial?
On Tue, 2005-01-25 at 20:57 +0100, Wilson Pickett wrote: L( ) option is applicable here? And, if your version of Asterisk doesn't have a Dial app with the L( ) option, will it be worth your while to upgrade to have the L( ) option? A third question might be in what version was this introduced. Don't know. I **THINK** it's been in there for a while. Several months at least. Isn't it in the 1.0 release? I've been using the HEAD version for a long time... I haven't been keeping track of versions murf signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] off topic - DECT phones with FSK VMWI in the UK
Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 Message Light on multiple phones
Here is what I am attempting to do (which works well on Cisco Call Manger). I have some 7960's that have multiple lines on them. The second line specifically is a helpdesk line that is shared among multiple phones. Here is how I am making that line ring on multiple phones, maybe you have other suggestions: exten = 135,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED],20,rt) So this rings the second line on the phones that have the first line as 100 and 101. This works great. When someone leaves a voicemail, the messagelight will only light on the phone that was booted up last. Is there a way to make the light come on all of the helpdesk phones, with the second line icon displaying the correct mail icon? Here is the sip.conf section for those particular extensions: [100] type=friend username=100 secret=100 host=dynamic mailbox=100 linelabel=First Last line = 102 [135] type=friend username=135 secret=135 host=dynamic mailbox=135 linelabel=HelpDesk line = 135 [101] type=friend username=101 secret=101 host=dynamic mailbox=101 linelabel=First1 Last1 callerid=First1 Last1 101 line = 101 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Telrad + EM T1 Trunk
All, One of our customers is using a Telrad PBX, we are providing phone server through asterisk via a T1 using em directly connected to the Telrad system. We're using a T1 cross cable as normal, the T1 part works great. No alarms. When we try and dial out the Telrad using a direct trunk group, the call fails. When looking in the asterisk console I noticed only 1 or 2 digits are seen, either that or there is some kind of timeout. Audio works fine because we hear Alice's no service message. Now for the details, initially we tried em_w (wink). zapata.conf is straight enough forward, immediate=no, we've tried messing with callprogress. We've tried playing with the wink/rxwink timings to no avail. It appears as if Asterisk is getting the digits and timing out too quickly. For kicks we put the Telrad into LCR mode, to dump the digits immediately that did nothing either. We tried em immediate also and that did nothing. Any suggestions? This is my first em turn up. At first I thought it was extensions.conf causing the problem. I have it simply as this: exten = _NXX,1,Dial(blah blah) exten = _NXXNXX,1,Dial(blah blah) exten = _1NXXNXX,1,Dial(yada yada) etc .. Thanks all, Matt S ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] off topic - DECT phones with FSK VMWI in the UK
On 26 Jan 2005, at 13:11, Chris Stenton wrote: Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas Can you post the relevant extension from sip.conf and the contents of voicemail.conf. Also, check your caller ID is set to the UK standard under Regional on your Sipura. Phil. -- Phil Quinney IT Consultant - Any-Ideas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Message Light on multiple phones
Here is what I am attempting to do (which works well on Cisco Call Manger). I have some 7960's that have multiple lines on them. The second line specifically is a helpdesk line that is shared among multiple phones. Here is how I am making that line ring on multiple phones, maybe you have other suggestions: exten = 135,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED],20,rt) So this rings the second line on the phones that have the first line as 100 and 101. This works great. When someone leaves a voicemail, the messagelight will only light on the phone that was booted up last. Is there a way to make the light come on all of the helpdesk phones, with the second line icon displaying the correct mail icon? I believe you'll find the phone that registered 'last' will be the one that gets the vm lite (not the last reboot). If your phones re-register ever 3600 seconds, the last one gets the mwi indicator and that will cause the mwi to move between phones over time. (Snom phones had a similar problem some time ago.) I believe the current implementation for vm notification is to use a sip 'notify' message to turn on the mwi, and the sip protocol implementation within * does not support sending 'notify' messages to multiple phones. (E.g., how would * even know how many phones you are trying to ring via the above dialplan entry?) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Am I missing something really basic here????? help with Asterisk@home
Im trying to install [EMAIL PROTECTED], Ive just downloaded the latest cd from soundforge. I can get it to install ok (network card didnt auto configure but I worked out how to use netconfig). I worked out how to add a few grandstream budgetone fine. Worked out how to upload music etc. Worked out how to modify FOP. Voicemail and meetmes work fine. HOWEVER. Im using a X100p. I cant get it to make a call out or use the default extension for an incoming line. What do I need to make the pstn connection work? Do I need to modify Zapata.conf? there are zero instructions on the [EMAIL PROTECTED] page as to what to do. Can anyone help me out here. TIA, Dean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Florz patch for zaphfc
Ivan Meic (Vox Mundi) wrote: Stuart, Can you plase specify in which mode are you using your hfc cards ? You said ptp, but are they working as NT or TE ? Ivan, I'm using the hfc cards in ptp mode connected to the pstn in TE-mode. During testing we used the same setup on a different isdn-line in ptmp mode witch equally good results. When we switched to the lines used by our old alcatel pbx I had wery strange problems until I realised that those lines where in PTP mode. /Nils Thanks, Ivan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stuart Hirst Sent: Wednesday, January 26, 2005 12:26 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Florz patch for zaphfc Nils, Thanks for your help with this issue and I thought I should send this to the list for the benefit of others. The issue was with the options piping through patch. The command I used was zcat /path/zaphfc_0.2.0-rc2b_florz-1.diff.gz | patch which worked fine. Notice no p1 option. The feedback from the customer is that the audio quality has improved 100% and they are now happy. I cannot find any errors from Asterisk or the system logs and so this patch from Florz seems to be good and stable. Thanks Florz. It is that good that where I have a 4 port BRI card from Junghanns in a site that is giving me major problems with framing errors where I am using three of the ports, I am going to install 3 Billion cards using zaphfc and the Florz patch to try and get around the problem because the framing error problems with the 4 port card is taking forever to resolve. Thanks again Nils for your help and Florz for a great patch. Rgds, Stuart -Original Message- From: Nils Segerdahl [mailto:[EMAIL PROTECTED] Sent: 23 January 2005 21:05 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Florz patch for zaphfc On Sun, 23 Jan 2005, Stuart Hirst wrote: Has anyone had any success using the Florz patch for zaphfc ? I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN lines however the users are complaining of crackles on the line which I am assuming is related to the IRQ issues raised by Florz. I have tried to use the patch but it errors trying to patch zaphfc.h Any help would be appreciated. Im running bristuff-0.2.0-rc2b with Florians patch. 4 Billion hfc cards in ptp mode. Works like a charm. Even spandsp for receiving faxes works. Pelase describe your problem in more detail. /Nils Nils Segerdahl --- Upsala Systemkonsult, UPSYS AB Telefon:(+46) (0)18 56 80 41 Glunten, 751 83 UppsalaMobil: (+46) (0)703 55 65 03 http://www.upsys.seFax: (+46) (0)18 56 80 49 --- Jan 24 Eskimo Pie patented by Christian Nelson, 1922 Jan 24 Gold discovered in California at Sutter's Mill, 1848 Jan 24 DG Nova introduced, 1969 --- -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nils Segerdahl Upsala Systemkonsult, UPSYS AB Telefon:(+46) (0)18 56 80 41 Upsala Science Park, 751 83 Upsala Mobil: (+46) (0)703 55 65 03 http://www.upsys.se Fax: (+46) (0)18 56 80 49 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7960
Hi I'm trying to deploy asterisk, but I can't seem to find documentation for the TFTP server to run the cisco 7960 ip phones' I was told before that you need it and it could run on linux. Thank You ,jm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7960
Hello! Doesn't matter which TFTP server you will setup. Any kind of TFTP will do it (Linux, Windows, Solaris, FreeBSD...). BR, Alen Jose Cruz (Branders IT) wrote: Hi I'm trying to deploy asterisk, but I can't seem to find documentation for the TFTP server to run the cisco 7960 ip phones' I was told before that you need it and it could run on linux. Thank You ,jm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960
Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7960 Hello! Doesn't matter which TFTP server you will setup. Any kind of TFTP will do it (Linux, Windows, Solaris, FreeBSD...). BR, Alen Jose Cruz (Branders IT) wrote: Hi I'm trying to deploy asterisk, but I can't seem to find documentation for the TFTP server to run the cisco 7960 ip phones' I was told before that you need it and it could run on linux. Thank You ,jm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi audio issue
no audio with echosquelch=0 in capi.conf can someone compile chan_capi changing the Makefile with CAPI_ES disabled and CAPI_GAIN enabled no audio in the channel I had to disable the CAPI_ES and CAPI_GAIN to get it working Can someone confirm this? Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Interesting Bellster issue
dhh When I make a call, bellster anounces that I have no credits and dhh says goodbye, but it still routes the call. Even if you have no credits, we'll try to route the call via DUNDi. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960
I'm not sure if this will work with your cisco's but I can guarantee that it works with the grandstreams. This is what I use to update my 4 phones, running on my main winxp machine and it's free for non commercial use. http://www.weird-solutions.com/product/tftpc2000.html Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jose Cruz (Branders IT) Sent: Wednesday, January 26, 2005 9:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960 Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7960 Hello! Doesn't matter which TFTP server you will setup. Any kind of TFTP will do it (Linux, Windows, Solaris, FreeBSD...). BR, Alen Jose Cruz (Branders IT) wrote: Hi I'm trying to deploy asterisk, but I can't seem to find documentation for the TFTP server to run the cisco 7960 ip phones' I was told before that you need it and it could run on linux. Thank You ,jm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] off topic - DECT phones with FSK VMWI in the UK
-Original Message- From: Chris Stenton [mailto:[EMAIL PROTECTED] Sent: 26 January 2005 13:12 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] off topic - DECT phones with FSK VMWI in the UK Off topic but I am after a DECT phone to connect to my sipura 3000 that has a FSK VMWI light or flashing envelope on the LCD screen. Any ideas Chris Possibley helpful but possibley not. These two phones work a treat with mediatrix (I would expect the same would be true with Sipura is the 3000 does support VMI), tho to access Vmail you need to manually enter a quick dial or I just haven't found the setting yet: BT Freestyle 2100 BT Calypso 1100 Colour However after buying two to test we found that the cheaper none colour version was much better audio quality so we went with the Freestyles instead. HTH alex This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960
Yes, this is frustrating I know. In fact the wiki could be updated to provide this info. Basically if you have the phones out of the box (brand spankin new) then you probly have the SCCP image installed on it by default. Your tftp server root will need a number of files to start if this is the case. Ok with that said, most of this I had to figure out on my own. Cisco's website as we all know is a pain in the a$$ to find any useful info on how to do anything. Be sure to remove #comments before experimenting. ALSO, DO THIS WITH ONE PHONE AT A TIME. If you have other phones plugged in they WILL automatically try to upgrade :) .. [OS79XX.txt] P0S3-07-3-00 # This is the version we used, S stands for Sip, 7-3 stands for 7.3 .. If you need firmware you'll have to get them off Cisco's site, there was a posting recently stating where to obtain these without a cisco login. # once you have that file in place your SEP (yes, SEP) device will start looking for the file with that extension. It cuts off the file extension, for example in your tftp root you will need: P0S3-07-3-00.sb2 P0S3-07-3-00.loads #Once you have those 3 files your phones should start upgrading, be careful though. It's been known that older versions that come on the phones have bugs and can blow up (crash) if you try to put too large an image on them. # Moving on, once you get that completed your phone should boot and start looking for the following files. Before I post them below, take note on how this all works. First you have a general config file, SIPDefault.cnf .. This contains such things as your proxy address, logo, services, directories, ntp, that kind of stuff. The second is your SIPMAC ADDRESS.cnf .. This is per phone, that contains your phone line info, names, etc.. [sipdefault.cnf] # Image Version image_version: P0S3-07-3-00 # Proxy Server proxy1_address: 192.168.1.17 # Proxy Server Port (default - 5061) #proxy1_port:5060 # NAT/Firewall Traversal nat_enable: 0 nat_address: voip_control_port: 5061 start_media_port: 16384 end_media_port: 32766 nat_received_processing: 0 # Proxy Registration (0-disable (default), 1-enable) proxy_register: 1 # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: 120 # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Enable VAD (0-disable (default), 1-enable) enable_vad: 0 # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable: 1 ; 0-Disabled, 1-Enabled (default) # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: 0 ; 0-Disabled, 1-Enabled (default) # Telnet Level (enable or disable the ability to telnet into this phone telnet_level: 1 ; 0-Disabled (default), 1-Enabled, 2-Privileged # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 11 sip_invite_retx: 6 ; Default 7 timer_invite_expires: 180; Default 180 sec # Setting for Message speeddial to UOne box messages_uri: 8500 #* Release 2 new config parameters ** # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: ./ # Time Server sntp_mode: directedbroadcast sntp_server: 17.254.0.49 time_zone: CST dst_offset: 1 dst_start_month: April dst_start_day: dst_start_day_of_week: Sun dst_start_week_of_month: 1 dst_start_time: 02 dst_stop_month: Oct dst_stop_day: dst_stop_day_of_week: Sunday dst_stop_week_of_month: 8 dst_stop_time: 2 dst_auto_adjust: 1 # Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control) dnd_control: 0 ; Default 0 (Do Not Disturb feature is off) # Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) callerid_blocking: 0; Default 0 (Disable sending all calls as anonymous) # Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control) anonymous_call_block: 0 ; Default 0 (Disable blocking of anonymous calls) # Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control) call_waiting: 1 ; Default 1 (Call Waiting enabled) # DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127) dtmf_avt_payload: 101 ; Default 100 # XML file that specifies the dialplan desired dial_template: dialplan # Network Media Type (auto, full100, full10, half100, half10) network_media_type: auto #Autocompletion During Dial (0-off, 1-on
RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960
Sorry my mistake, wrong link, here is the correct one. http://www.weird-solutions.com/product/tftp-desktop.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: Wednesday, January 26, 2005 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960 I'm not sure if this will work with your cisco's but I can guarantee that it works with the grandstreams. This is what I use to update my 4 phones, running on my main winxp machine and it's free for non commercial use. http://www.weird-solutions.com/product/tftpc2000.html Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jose Cruz (Branders IT) Sent: Wednesday, January 26, 2005 9:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960 Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7960 Hello! Doesn't matter which TFTP server you will setup. Any kind of TFTP will do it (Linux, Windows, Solaris, FreeBSD...). BR, Alen Jose Cruz (Branders IT) wrote: Hi I'm trying to deploy asterisk, but I can't seem to find documentation for the TFTP server to run the cisco 7960 ip phones' I was told before that you need it and it could run on linux. Thank You ,jm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960
Jose Cruz (Branders IT) wrote: But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? Jose, Under Mandrake, to install the tftp program is, urpmi tftp-server. The tftp root directory is, /var/lib/tftproot. It will vary under different distros. Once you get your tftp server running, you need to copy the 7960's SIP image you got from the Cisco website into the tftp root and follow their instructions on how to apply to the phone. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC-S card problems
Hello everybody,
I'm having little trouble (well, pretty big trouble) with HFC-S card and
Asterisk. My idea is to do VoIP/IAX link between two HW PBXen using two
Asterisk PC boxen with ISDN cards in them. AFAIK HFC-S cards must be in NT
mode for this installation, they must behave like state line for those HW
PBXen. (if wrong, please correct me).
Diagram follows:
Internet
Phone1-PBX1-ISDN[NT]-ASTERISK1--IAX--ASTERISK2-[NT]ISDN-PBX2-Phone2
The problem is:
I don't even know if HFC-S cards are working. Module is loaded fine,
everything seems OK but when I try to dial somewhere using Asterisk CLI
nothing happens. (well it says no answer, but HW PBX doesn't even know
that somebody is trying to dial - I've got sysphone with those nice LEDs
and none of them is emiting light :)
CLI dial 220
-- Executing Dial(OSS/dsp, Zap/1/36|10) in new stack
-- Called 1/36
-- Channel 0/1, span 1 got hangup
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
When I do fast show channels, this is the result:
CLI show channels
Channel (ContextExtensionPri ) State Appl. Data
Zap/1-1 (test s1 ) Dialing AppDial (Outgoing Line)
OSS/dsp (test 220 1 ) Ringing Dial Zap/1/36|10
My Questions:
Is there any application which can just dial somewhere or at least run
some data through line? I just need to test that there's dataflow between
HFC-S and PBX. I need something without hard setup, just simple test tool.
Is there anybody who would be so kind to send me his/hers
Asterisk/Zapata/Zaptel config files so I can check them with mine? I know,
configs are somewhat personal stuff, but it will _very_ appreciated :)
Here come all necessary information from my config files. Don't take them
too seriosly, it's just a test config with no actual usage. BIG TNX in
advance for at least some help:
zaptel.conf
---
defaultzone=nl
loadzone=nl
span=1,1,3,ccs,ami
bchan=1,2
dchan=3
syslog
--
PCI: Enabling device 00:0f.0 ( - 0003)
zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xca8fd000 fifo
0xc6348000(0x6348000) IRQ 11 HZ 100
zaphfc: Card 0 configured for NT mode
Registered Span 1 ('ZTHFC1') with 3 channels
Span ('ZTHFC1') is new master
zaphfc: 1 hfc-pci card(s) in this box.
ztcfg -vv
-
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
3 channels configured.
zapata.conf
---
[channels]
language=en
switchtype = euroisdn
pridialplan=unknown
prilocaldialplan=unknown
signalling = bri_net_ptmp
echocancel=yes
immediate=no
group = 1
context = test
channel = 1
channel = 2
extensions.conf
---
[test]
exten = 210,1,Wait(1)
exten = 210,2,Answer
exten = 210,3,Playback(demo-congrats)
exten = 210,4,Hangup
exten = 220,1,Dial(Zap/1/36,10)
Ufff. Hope that's it...
Big thanx once more...
--ZK
- - ---[ CESKE TELEKOMUNIKACE ]-- - -
Zdik Kudrle
GSM: +420 604 781 414
HTTP: www.cesketelekomunikace.cz
SMTP: [EMAIL PROTECTED]
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RE: [Asterisk-Users] Cisco 7960 Message Light on multiple phones
That's very interesting, because we do the exact same thing and all the phones light up (with line mailbox flashing).. What SIP ver are you using on the 7960's? However it sounds like 135 isn't registered on all the phones? What we did is bind the lines to multiple phones, 203 (our tech mailbox) for example never actually rings because all we care about is the MWI. Matt -Original Message- From: Mark Johnson [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 26, 2005 7:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 Message Light on multiple phones Here is what I am attempting to do (which works well on Cisco Call Manger). I have some 7960's that have multiple lines on them. The second line specifically is a helpdesk line that is shared among multiple phones. Here is how I am making that line ring on multiple phones, maybe you have other suggestions: exten = 135,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED],20,rt) So this rings the second line on the phones that have the first line as 100 and 101. This works great. When someone leaves a voicemail, the messagelight will only light on the phone that was booted up last. Is there a way to make the light come on all of the helpdesk phones, with the second line icon displaying the correct mail icon? Here is the sip.conf section for those particular extensions: [100] type=friend username=100 secret=100 host=dynamic mailbox=100 linelabel=First Last line = 102 [135] type=friend username=135 secret=135 host=dynamic mailbox=135 linelabel=HelpDesk line = 135 [101] type=friend username=101 secret=101 host=dynamic mailbox=101 linelabel=First1 Last1 callerid=First1 Last1 101 line = 101 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960
you'll have to get them off Cisco's site, there was a posting recently stating where to obtain these without a cisco login. Anyone care to remind us of where this was??? (did a quick search and didn't see anything immediately). Since we are outside the US, Cisco US refuse to 'sell' us a login and to speak to Cisco UK, Cisco UK tell us to contact a re-supplier, and all re-suppliers I have spoken to haven't got a clue as to what is the European equivalent part number to the US one, therefore shaking their heads and telling us that they cant supply. I know how it SHOULD be done, but they make it near-on impossible :( :( Got fed up going round in circles in the end. all for $8 worth of access :( Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ParkAndAnnounce +${ALERT_INFO}
I am trying to use the _ALERT_INFO variable with ParkAndAnnounce. My idea was to have the phone, a Polycom IP500 auto answer so you could hear the annoucement of the parked extension over the speaker. This variable works fine with the normal Dial application, but seems to be ignored by ParkAndAnnounce. I am not knowledgable enough to know if this is normal operation, but a syntax error at my side. Also, is it possiple to include multiple SIP extensions in ParkAndAnnounce just as in the Dial application. I tried the SIP/1001SIP/1002 context, but it was interpreted as a bad extension by ParkAndAnnounce. Thanks. Ed Horton ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Howto Setup TFTP server on Linux for Cisco 7960
Get a TFTP for Linux if you use Red Hat or Fedora Core get the server there : http://dag.wieers.com/packages/tftp/ it's also available on the install CD... Then to know what file you need to change your Cisco 7960 phone from Skinny to SIP go to this website as it explain how to do it. If it doesn't work let me know I did it 2 weeks ago. You have to start with a SIP image of version 2.0 or 2.1 if you want it to work. Then you can upgrade to 2.2, then 3.x, then 4.x, etc. all the way up to 7.3. For me I was able to do it that way : Started with SIP 2.1 then 3.3 then 6.3 and then 7.3. But I had a hard time upgrading from 2.1 to 3.3 as the 3.3 image as more then 8 characters and firmware 2.1 support only 8:3 (8 characters plus 3 characters for the extension) If you need any help let me know. Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960
Got fed up going round in circles in the end. all for $8 worth of access :( Technically, Cisco wants you to pay for those images :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Message Light on multiple phones
Rich Adamson wrote: Here is what I am attempting to do (which works well on Cisco Call Manger). I have some 7960's that have multiple lines on them. The second line specifically is a helpdesk line that is shared among multiple phones. Here is how I am making that line ring on multiple phones, maybe you have other suggestions: exten = 135,1,Dial(SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED],20,rt) So this rings the second line on the phones that have the first line as 100 and 101. This works great. When someone leaves a voicemail, the messagelight will only light on the phone that was booted up last. Is there a way to make the light come on all of the helpdesk phones, with the second line icon displaying the correct mail icon? I believe you'll find the phone that registered 'last' will be the one that gets the vm lite (not the last reboot). If your phones re-register ever 3600 seconds, the last one gets the mwi indicator and that will cause the mwi to move between phones over time. (Snom phones had a similar problem some time ago.) I believe the current implementation for vm notification is to use a sip 'notify' message to turn on the mwi, and the sip protocol implementation within * does not support sending 'notify' messages to multiple phones. (E.g., how would * even know how many phones you are trying to ring via the above dialplan entry?) I was hoping that asterisk would be able to sort that out. The neatest part about this setup is that this shared extension can have multiple calls going on. Example: on Cisco Call Manger if you have a shared extension between three phones and someone picks up the line, none of the other phones can use that extension. With SIP, If the same person picks up the line, so can the other two people. The message light working on all of the phones would be great! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960
Got fed up going round in circles in the end. all for $8 worth of access :( Technically, Cisco wants you to pay for those images :) Indeed, and I would if Cisco made it Technically possible! :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7960
I'm trying to deploy asterisk, but I can't seem to find documentation for the TFTP server to run the cisco 7960 ip phones' I was told before that you need it and it could run on linux. Try 'man tftpd' at your favorite unix command line. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Howto Setup TFTP server on Linux for Cis co 7960
Martin What website? I think you forgot to put the link to the site on how to do it. Thanks ,jm -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Roy Sent: Wednesday, January 26, 2005 6:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Howto Setup TFTP server on Linux for Cisco 7960 Get a TFTP for Linux if you use Red Hat or Fedora Core get the server there : http://dag.wieers.com/packages/tftp/ it's also available on the install CD... Then to know what file you need to change your Cisco 7960 phone from Skinny to SIP go to this website as it explain how to do it. If it doesn't work let me know I did it 2 weeks ago. You have to start with a SIP image of version 2.0 or 2.1 if you want it to work. Then you can upgrade to 2.2, then 3.x, then 4.x, etc. all the way up to 7.3. For me I was able to do it that way : Started with SIP 2.1 then 3.3 then 6.3 and then 7.3. But I had a hard time upgrading from 2.1 to 3.3 as the 3.3 image as more then 8 characters and firmware 2.1 support only 8:3 (8 characters plus 3 characters for the extension) If you need any help let me know. Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec negotiation
Can't you just create a different context for inbound and outbound calls? Then specify your codec preference order in there. I don't think you can specify the bandwidth= parameter within a context other than the global one though. -mark On Jan 25, 2005, at 6:13 PM, [EMAIL PROTECTED] wrote: I don't want that... because - for outbound calls I want priority to be g729 first - for inbound calls I want no priority at all (e.g. the calling asterisk to decide which codec we will use) The last doesn't happen.. This by the way DID happen correctly with previous versions of asterisk (1.0.3 for example) the current CVS-HEAD version doesn't -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohammed Salim Sent: dinsdag 25 januari 2005 22:10 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Codec negotiation The order matters in asterisk so if you want GSM to take priority over G729, simply put that ahead of the G729... so your settings should be: Allow=all Allow=gsm Allow=g729 Allow=ulaw Allow=alaw Try that and see if it works. Regards, Mohammed Salim EZZI Telecom, Inc. -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 600 - 1.3.1
I am getting to my wits end with these phones (and so is my boss). I am getting an random echo on these phones and I have an issue opened with Polycom and its been in their research and development department for almost a month with no results. I have noticed that I get a message RFC3389 support incomplete. Turn off on client if possible in asterisk. I have researched this and made the change in ipmid.cfg (see below), but I am still getting this RFC error. --- ipmid.cfg RTP qos.ethernet.rtp.user_priority=5/ RTP qos.ip.rtp.min_delay=0 qos.ip.rtp.max_throughput=0 qos.ip.rtp.max_reliability=0 qos.ip.rtp.min_cost=0 qos.ip.rtp.precedence=0/ RTP tcpIpApp.port.rtp.filterByIp=1 tcpIpApp.port.rtp.filterByPort=0 tcpIpApp.port.rtp.forceSend= tcpIpApp.port.rtp.mediaPortRangeStart=/ - end I am just wondering if anyone can help me troubleshoot the echo and RFC error so I don't have to pull the entire phone system out and purchase an entire new system. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF digit dropping
I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option related to DTMF). Anyone else getting similar drops? Any solutions? Is http://connect.voicepulse.com/ , using IAX, any better? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF digit dropping
I'm having the same problem with Voicepulse connect using IAX2. So, no, it's not better IMHO. And I've been thinking about switching to SIP to see if the problem goes away (I'm very reluctant to do so though) but it's hard to know if the problem lies with Voicepulse (or Broadvoice in your case) or whatever CLEC terminates your inbound number. FWIW I have experienced the problem with Asterisk 1.0.2 and now also with 1.0.5. It doesn't seem to be an Asterisk problem though because the vast majority must not be having any issues with DTMF recognition. -mark On Jan 25, 2005, at 8:55 PM, Bryce Nesbitt (mailing list account) wrote: I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option related to DTMF). Anyone else getting similar drops? Any solutions. Is http://connect.voicepulse.com/ , using IAX, any better? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rining Issues
When I access the Directory() and use it to call an extension, the origination hears garbled or inconsistent ringing. The termination side rings normally and the conversation is clean in both directions. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone having problems with LiveVoIP?
It would be really nice to see whatever patches they develop for Asterisk or at least get some hint of where the problem lies. -mark On Jan 25, 2005, at 9:41 PM, Brian Dingman wrote: LiveVoip has a problem with Asterisk users on versions less than 1.0.3 If you are not using that version you need to upgrade now. We have a problem with two of our carriers at their gateway related to the Asterisk users. Our staff has developed a patch that is being tested at this time. Once the patch has been approved on our testbed we will move it on to the production switch environment. We do not do upgrades like this during the hours or 9 a.m. - 7 p.m. EST due to high traffic loads. We expect to do switch updates after 7 p.m. this evening that should resolve the problems you are having. LiveVoip engineers are also looking at a DTMF problem in the Asterisk software ver. 1.0.3 which may or may not involve you. Both of these issues are Asterisk software related in nature and not LiveVoip LLC switching defects. Thank You in Advance for your understanding. This issue has been placed under a master ticket for tracking. ** When contacting LiveVoip LLC Support please provide us with the latest version of Asterisk you are using, any and all logs if necessary and as much detail regarding any problems you are having. Network Operations Team LiveVoip LLC On Tue, 25 Jan 2005 20:11:43 -0600, Tim Lewis [EMAIL PROTECTED] wrote: Thanks Jeff! I think it's a little too late to find this info out. 3 to 4 days of no service. I have send many emails and still awaiting a response. Reminds me of my ILEC (QWEST) Do you have any info on what this patch does? -later On Tue, 2005-01-25 at 18:20, Jeff Glassman wrote: They are coming out with a patch for the DID problem tonight. Need to have Asterisk 1.0.3 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF digit dropping
Is it not possible to use sip debug or ethereal to see what digits arrive at your site? (or do you already know the digits are mutilated before getting to you?) I'm having the same problem with Voicepulse connect using IAX2. So, no, it's not better IMHO. And I've been thinking about switching to SIP to see if the problem goes away (I'm very reluctant to do so though) but it's hard to know if the problem lies with Voicepulse (or Broadvoice in your case) or whatever CLEC terminates your inbound number. FWIW I have experienced the problem with Asterisk 1.0.2 and now also with 1.0.5. It doesn't seem to be an Asterisk problem though because the vast majority must not be having any issues with DTMF recognition. -mark On Jan 25, 2005, at 8:55 PM, Bryce Nesbitt (mailing list account) wrote: I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option related to DTMF). Anyone else getting similar drops? Any solutions. Is http://connect.voicepulse.com/ , using IAX, any better? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with Grandstream ringback
My phones were running firmware version x.18. There was a field that allowed me to select automatic updates and how often. I selected yes and set it to 1 day. (I thought maybe 0 days would cause it to update immediately, but all it caused was an error.) There was a field for http updates. I set it to yes and set the http address to http://fm.grandstream.com/gs/ I then powered down the phone and powered it back up. This caused the firmware to upgrade. I then logged into it via the web interface and checked the firmware version on the basic tab. It was then at x.22. I say was in the above statement because the update fields are arranged a bit differently in x.22 and I am going from memory when I speak of the x.18 fields. On Wed, 2005-01-26 at 09:00 +0100, Robert Rozman wrote: - Original Message - From: Kim Lux [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 26, 2005 8:13 AM Subject: Re: [Asterisk-Users] Asterisk with Grandstream ringback I updated to firmware version x.22 and this and a few other problems were fixed. I was running x.18 and it allowed me to do a successful upgrade via http. Hi, could you please post your settings for http upgrade and url for firmware ? Thanks, Rob. On Tue, 2005-01-25 at 08:10 -0700, Kim Lux wrote: Are you saying that you are running firmware X.22 and it is not doing the callback when you hang up ? Where exactly did you get that firmware version ? Thanks On Tue, 2005-01-25 at 16:55 +0200, Doug Reid - Stormcorp wrote: Hi All Has any one tested Ver X.22 on the grandstreams? If so have you noticed the problem experienced with ringback? When you hang up the GS rings again and its the same call you put down. Only seen this with Ver X.16 and X.18 not yet with X.22 but I'm still not 100% convinced. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kim Lux, Diesel Research Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kim Lux, Diesel Research Inc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone having problems with LiveVoIP?
Just to let everyone using [EMAIL PROTECTED] know that my livevoip DID now works without any changes to asterisk! Jeff -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Eissler Sent: Wednesday, January 26, 2005 10:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Brian Dingman Subject: Re: [Asterisk-Users] Anyone having problems with LiveVoIP? It would be really nice to see whatever patches they develop for Asterisk or at least get some hint of where the problem lies. -mark On Jan 25, 2005, at 9:41 PM, Brian Dingman wrote: LiveVoip has a problem with Asterisk users on versions less than 1.0.3 If you are not using that version you need to upgrade now. We have a problem with two of our carriers at their gateway related to the Asterisk users. Our staff has developed a patch that is being tested at this time. Once the patch has been approved on our testbed we will move it on to the production switch environment. We do not do upgrades like this during the hours or 9 a.m. - 7 p.m. EST due to high traffic loads. We expect to do switch updates after 7 p.m. this evening that should resolve the problems you are having. LiveVoip engineers are also looking at a DTMF problem in the Asterisk software ver. 1.0.3 which may or may not involve you. Both of these issues are Asterisk software related in nature and not LiveVoip LLC switching defects. Thank You in Advance for your understanding. This issue has been placed under a master ticket for tracking. ** When contacting LiveVoip LLC Support please provide us with the latest version of Asterisk you are using, any and all logs if necessary and as much detail regarding any problems you are having. Network Operations Team LiveVoip LLC On Tue, 25 Jan 2005 20:11:43 -0600, Tim Lewis [EMAIL PROTECTED] wrote: Thanks Jeff! I think it's a little too late to find this info out. 3 to 4 days of no service. I have send many emails and still awaiting a response. Reminds me of my ILEC (QWEST) Do you have any info on what this patch does? -later On Tue, 2005-01-25 at 18:20, Jeff Glassman wrote: They are coming out with a patch for the DID problem tonight. Need to have Asterisk 1.0.3 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS for Asterisk
-Original Message- From: Michael 'Moose' Dinn [mailto:[EMAIL PROTECTED] You should be able to boot Asterisk using slackware as a base from a 64M CF card or even from a 64M bootable USB memory key. If you use ReiserFS or something similar for the drive that stores all your voicemail, etc then it should come back without a problem as well. Of course you want to make sure the system shuts down cleanly too... Or use a journalling filesystem. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960
I use it for my 7960's at the house and it works fine. dean collins wrote: I'm not sure if this will work with your cisco's but I can guarantee that it works with the grandstreams. This is what I use to update my 4 phones, running on my main winxp machine and it's free for non commercial use. http://www.weird-solutions.com/product/tftpc2000.html Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jose Cruz (Branders IT) Sent: Wednesday, January 26, 2005 9:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960 Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7960 Hello! Doesn't matter which TFTP server you will setup. Any kind of TFTP will do it (Linux, Windows, Solaris, FreeBSD...). BR, Alen Jose Cruz (Branders IT) wrote: Hi I'm trying to deploy asterisk, but I can't seem to find documentation for the TFTP server to run the cisco 7960 ip phones' I was told before that you need it and it could run on linux. Thank You ,jm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZT_CHANCONFIG failed on channel 11: Function not implemented (38)
I am running RedHat Fedora Core 3 with their kernel-2.6.10-1.741_FC3 kernel, and if I use nethdlc in zaptel.conf, when I run ztcfg I get the message: ZT_CHANCONFIG failed on channel 11: Function not implemented (38) Does anyone have ideas on how I can work around this issue? T1 channels 1-10 are voice, 11-24 are data (second board isn't used at the moment). Here is my /etc/zaptel.conf file: span=1,1,5,esf,b8zs span=2,0,0,esf,b8zs fxoks=1-10 nethdlc=11-24 #clear=11-24 clear=25-48 loadzone = us defaultzone=us ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Howto Setup TFTP server on Linux for Cisco 7960
Oops forgot to give the second link... here it is : http://www.wheely-bin.co.uk/cisco/ For people requesting SIP images for Cisco phones there's a few link where you can get them here's 2 I found : http://www.m4tr1xx.de/cisco/ (there's SIP 6.3, 7.0 and 7.1 there) http://ns.goodgrief.com/voice-comm/ (there's 2.x,3.x and 4.x there but the 4 didn't seem to work for me) As weird as it might seem you can also find some on eDonkey and Kazaa if you are willing to wait a day or so to download them... Hope this help Martin Roy Martin Roy wrote: Get a TFTP for Linux if you use Red Hat or Fedora Core get the server there : http://dag.wieers.com/packages/tftp/ it's also available on the install CD... Then to know what file you need to change your Cisco 7960 phone from Skinny to SIP go to this website as it explain how to do it. If it doesn't work let me know I did it 2 weeks ago. You have to start with a SIP image of version 2.0 or 2.1 if you want it to work. Then you can upgrade to 2.2, then 3.x, then 4.x, etc. all the way up to 7.3. For me I was able to do it that way : Started with SIP 2.1 then 3.3 then 6.3 and then 7.3. But I had a hard time upgrading from 2.1 to 3.3 as the 3.3 image as more then 8 characters and firmware 2.1 support only 8:3 (8 characters plus 3 characters for the extension) If you need any help let me know. Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960
On Wednesday 26 January 2005 14:34, Paul Brock wrote: you'll have to get them off Cisco's site, there was a posting recently stating where to obtain these without a cisco login. Anyone care to remind us of where this was??? (did a quick search and didn't see anything immediately). Since we are outside the US, Cisco US refuse to 'sell' us a login and to speak to Cisco UK, Cisco UK tell us to contact a re-supplier, and all re-suppliers I have spoken to haven't got a clue as to what is the European equivalent part number to the US one, therefore shaking their heads and telling us that they cant supply. I know how it SHOULD be done, but they make it near-on impossible :( :( Got fed up going round in circles in the end. all for $8 worth of access :( Try http://www.s2s.ltd.uk/ I'm just a satisfied customer. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy
I have my asterisk box showing an incoming call in the logs. But I get a message on the phone that the number is busy and I have to leave a message. I found out that the message was on my broadvoice voicemail. This happens everytime. I also saw something in the logs that says congestion for each incoming call. Any ideas how to fix this? Thanks! Randy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF digit dropping
Are you kidding? DTMF handling in Asterisk is the worst, it might not just be Asterisk. Dropped digits is a common thing. INBAND basically plays the sound of the DTMF digit, that's why G711 is the only codec it'll work on, it's the only high enough quality codec to play a touchtone and it be delivered correctly on the other side. We get the occasional dropped DTMF with Lookieloo, NuFone, VoipJet and BroadVoice. Across multiple servers running multiple Asterisk versions. However, I know that the later Asterisk versions (Past month or so) fixed some of the DTMF troubles we've had. The only thing I can suggest is try running the latest stable w/ BroadVoice, and maybe try rfc2833 I have a small IVR on my Asterisk server connected to BroadVoice, I always used DTMF, but I tried to switch to rfc2833 the other day out of curiosity and interesting enough, when I called into my IVR w/ my cell phone, it recognized 1234 and whatever other digits I entered. So inbound DTMF worked using ULaw, however I never tried outbound. Could have been a fluke though. Give it a shot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, January 26, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DTMF digit dropping Is it not possible to use sip debug or ethereal to see what digits arrive at your site? (or do you already know the digits are mutilated before getting to you?) I'm having the same problem with Voicepulse connect using IAX2. So, no, it's not better IMHO. And I've been thinking about switching to SIP to see if the problem goes away (I'm very reluctant to do so though) but it's hard to know if the problem lies with Voicepulse (or Broadvoice in your case) or whatever CLEC terminates your inbound number. FWIW I have experienced the problem with Asterisk 1.0.2 and now also with 1.0.5. It doesn't seem to be an Asterisk problem though because the vast majority must not be having any issues with DTMF recognition. -mark On Jan 25, 2005, at 8:55 PM, Bryce Nesbitt (mailing list account) wrote: I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option related to DTMF). Anyone else getting similar drops? Any solutions. Is http://connect.voicepulse.com/ , using IAX, any better? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Message Light on multiple phones
I believe the current implementation for vm notification is to use a sip 'notify' message to turn on the mwi, and the sip protocol implementation within * does not support sending 'notify' messages to multiple phones. (E.g., how would * even know how many phones you are trying to ring via the above dialplan entry?) This is interesting because I am doing a very similar thing. I have four Cisco phone, two 7940 and two 7905 and a couple of ata186. An incoming call rings all six phones. There is a single voicemail box that is assicate with every phone. The MWI indicator lights up on all phones when a message is received. It also extiguishes from all phones if the voicemail is deleted from any phone. Adi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone having problems with LiveVoIP?
LiveVoIP did not issue any end user patches last night. They had a problem connecting to Level 3's network. LiveVoIP claimed the problem was with asterisk users, I have not upgrade or install any patches and all is fine now. My main problem with LiveVoIP has been the LACK of customer service. They don't answer the phones or responded to email in a timely manner. How hard would it had been to post a message about the outage? On Wed, 2005-01-26 at 08:59, Mark Eissler wrote: It would be really nice to see whatever patches they develop for Asterisk or at least get some hint of where the problem lies. -mark On Jan 25, 2005, at 9:41 PM, Brian Dingman wrote: LiveVoip has a problem with Asterisk users on versions less than 1.0.3 If you are not using that version you need to upgrade now. We have a problem with two of our carriers at their gateway related to the Asterisk users. Our staff has developed a patch that is being tested at this time. Once the patch has been approved on our testbed we will move it on to the production switch environment. We do not do upgrades like this during the hours or 9 a.m. - 7 p.m. EST due to high traffic loads. We expect to do switch updates after 7 p.m. this evening that should resolve the problems you are having. LiveVoip engineers are also looking at a DTMF problem in the Asterisk software ver. 1.0.3 which may or may not involve you. Both of these issues are Asterisk software related in nature and not LiveVoip LLC switching defects. Thank You in Advance for your understanding. This issue has been placed under a master ticket for tracking. ** When contacting LiveVoip LLC Support please provide us with the latest version of Asterisk you are using, any and all logs if necessary and as much detail regarding any problems you are having. Network Operations Team LiveVoip LLC On Tue, 25 Jan 2005 20:11:43 -0600, Tim Lewis [EMAIL PROTECTED] wrote: Thanks Jeff! I think it's a little too late to find this info out. 3 to 4 days of no service. I have send many emails and still awaiting a response. Reminds me of my ILEC (QWEST) Do you have any info on what this patch does? -later On Tue, 2005-01-25 at 18:20, Jeff Glassman wrote: They are coming out with a patch for the DID problem tonight. Need to have Asterisk 1.0.3 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callmanager and Asterisk problem
Edgar de Leon wrote: Hello everybody, i got and asterisk and a CCM configured thru SIP, and in the sip show peers appear Name/usernameHostDyn Nat ACL Mask Port Status CCM 10.60.27.138255.255.255.255 5060 OK (1 ms) Just a guess, but do you have a username setup in sip.conf? It might be that or it might be the fromuser setting. Hey, lets see your sip.conf please? -Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Restart in the DISA to the beginning
Hi, Is there a way to restart the DISA to the enter phone number? For instance, Bell Calling Cards let you hit # at any point which lets you enter another number to call. This is useful to reduce the number of digits dialed and to utilize per-minute calls. I was not able to find anything on the web. Thanks, -Ryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BroadVoice Or VoicePulse ? {Scanned}
I have two Vonage lines and one Lingo line. I would like to drop them and go with BroadVoice. I would need to have three to four lines from BV. I should be able to configure Asterisk to handle all the SIP connections? Right Thanks, David PS its a home PBX. On Tue, 2005-01-25 at 11:39, Jay Milk wrote: From my experience, Broadvoice is good except for customer service. VoicePulse I wasn't too impressed with. Determine what your needs are, and see if a per-minute plan would work better for you. You can have US calls for 1.3c/minute incoming and outgoing if you shop around. -Original Message- From: Manjit Riat [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 25, 2005 12:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] BroadVoice Or VoicePulse ? Which would you recommend as far and quality and pricing to connect to asterisk (including DTMF issues)/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Interesting bellster issue
However, the second route tried by bellster ended up with This is 9:25 local time, calls are only permitted from ... to It means that the remote asterisk accepted the call to play the message, instead of using Congestion to use another route or fail. The playback command has a no answer option that can be used to play announcements without completing the call (ie: without providing answer supervision). It's important that people use this option correctly; not just for bellster, but for any error announcements. The example I always like to give is what happens to that quarter that someone at a payphone uses to call your errored number. Do they get it back? Or do they lose it just because they heard an announcement. It's pretty neat, even works well with the PSTN. I can call from my cell phone to a congested number, hear an announcement and my cell phone thinks the call was never completed. I'd assume the bellster network would be able to determine if the call was successfully connected or not. -Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setup questions- many users, little use
Bill Lattner wrote: Right now our setup is looking as follows: 12 ch T1 with 60 or 80 DIDs (using Digium T100P) P4 2.8/3 GHz Server (possibly a Dell 750, only running Asterisk) 41 Sipura ATAs (SPA-1001) 41 Sipura ATA's? Wow. that's a lot.. Have you thought about using channel banks or the Mediatrix product? Of course, if this is 1 ata per room, that's a different story. Just thinking that you could install the mediatrix in the building's closets and utilize the existing POTs cabling.. Just a thought.. 41 ATAs might be a pain to manage unless you are setup and prepared for that.. -Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] optimumvoice
I used them for a while and it was great. The service went down twice (once for an unknown reason for about 30 minutes and once on Christmas due to call volume). Call quality was excellent and they provided CNAM. It was all I needed. On Wed, 2005-01-26 at 13:03 +0200, Shoval Tomer wrote: Hi. A friend of mine came asking about this VOIP provider. I havent heard of them so I thought I might ask the list. Anyone has any experience using them (http://www.optonline.net/Home, http://www.optimumvoice.com/index.jhtml)? Not just with Asterisk, even with their supplied HandyTone and a regular analog phone. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Interesting bellster issue
However, the second route tried by bellster ended up with This is 9:25 local time, calls are only permitted from ... to It means that the remote asterisk accepted the call to play the message, instead of using Congestion to use another route or fail. I lost one credit without having the call placed, but what is more The owner of this node has fixed the problem. /ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Samuel Tardieu Sent: Wednesday, January 26, 2005 3:30 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Interesting bellster issue dhh == dhickman [EMAIL PROTECTED] writes: dhh When I make a call, bellster anounces that I have no credits and dhh says goodbye, but it still routes the call. I just noticed another interesting problem: I checked that using Congestion I can appropriately reject an incoming bellster call and that another route is used (on extension +331, France, Paris). However, the second route tried by bellster ended up with This is 9:25 local time, calls are only permitted from ... to It means that the remote asterisk accepted the call to play the message, instead of using Congestion to use another route or fail. I lost one credit without having the call placed, but what is more important is that no other route has been tried, and that my PBX thinks that the call succeedeed and will not try an alternative route such a Zap line. The problematic route is 179. Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Issue, Grandstream Sip.
I have a problem, when in a call with a grandstream phone it disconects around 4 mins into a conversation, I have tryed it with different providers, and still get the same results. But is wierd, in the grandstream phone i get a fast busy, while on the other side the call is still stablished AND even if the grandstream is giving fast busy if the person says something the other side can hear it, but all that the grandstream side hears is a busy tone, Thanks For Your Input, Albert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 Message Light on multiple phones
Adi Linden wrote: I believe the current implementation for vm notification is to use a sip 'notify' message to turn on the mwi, and the sip protocol implementation within * does not support sending 'notify' messages to multiple phones. (E.g., how would * even know how many phones you are trying to ring via the above dialplan entry?) This is interesting because I am doing a very similar thing. I have four Cisco phone, two 7940 and two 7905 and a couple of ata186. An incoming call rings all six phones. There is a single voicemail box that is assicate with every phone. The MWI indicator lights up on all phones when a message is received. It also extiguishes from all phones if the voicemail is deleted from any phone. Adi ___ Could you describe how you do that! That's exactly what I am trying to do! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Softphone
Hi, Does someone know an ActiveX IAX softphone? I need a free softphone to connect with Asterisk from a web page. Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
