Re: [asterisk-users] ip phone suggestion for Asia?
Hi tbskyd, We have found that the Grandstream's are not that great a phone. One of our best sellers is the Snom range and I know that the Australian supplier spends half his time in Hong Kong so you shouldn't have any problems getting so over there. They are a little more expensive than the Grandstream's but cheaper than the Polycoms around that Aastra price range. Cheers, Joel. On Thu, 2007-12-20 at 12:33 +0800, d tbsky wrote: Hi: i am surveying ip phones for our company. we will use them with asterisk. we have office in taiwan, hong kong,singapore and china. cisco and polycom are too expensive for us. we try several china brand ip phones. they are all cheap and some of them have good quality. but most of them won't offer future firmware support, which we think it's important for ip phones. searching in the mail list, we found aastra is good, but they don't sale to asia. grandstream looks good also.there are many grandstream users in the list, can someone share any good or bad experience about grandstream today? if there are other good choice, please tell us!! thanks a lot for your help!! Regards, tbskyd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mystery phone!
Hmm the shape looks like an Aastra but the buttons down the side look like PlayStation buttons to me. Maybe it's a Sony Cisco joint effort. Joel. On Mon, 2007-10-29 at 16:35 -0500, Kyle Sexton wrote: Does anyone know who really makes this phone: http://www.hybsys.bg/Products/VoIP/IP/Phones/5000/ Large pictures are at the bottom: http://www.hybsys.bg/img/ipph/IP5000_1.jpg http://www.hybsys.bg/img/ipph/IP5000_2.jpg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] centos 5 vs OpenSuse 10.3
We used to use CentOS 4 here but about 6-8 months ago we found that they were too slow with updates their repos for some of the 3rd party software that we were developing. We switched to SuSe 10.2 and haven't looked back. However Asterisk works equally well on both. Just pick your favorite flavor. Cheers, Joel. On Thu, 2007-10-18 at 11:34 -0500, Brian West wrote: I'm sorry I call bullshit on this one. CentOS has been 2.6 for some time. /b On Oct 18, 2007, at 11:22 AM, [EMAIL PROTECTED] wrote: Just 5 months ago CENTOS started to use Linux 2.6 one of the reasons I'd abandoned for SuSE a while back. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Music On Hold
Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? Cheers, Joel. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
Thanks for the suggestion, but I need it to play multiple messages. Always starting with the same one. Cheers, Joel. On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote: Make the file the only one in the /var/lib/asterisk/moh directory. Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 26, 2007, at 3:07 AM, Joel Hill wrote: Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? Cheers, Joel. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: DELL Platforms
Hi, About 2 years ago we made the decision to ship exclusively Dell servers. Mostly we have shipped the 860 rackmount with a config of a basic dual core proc couple gig of RAM and a pair of 75GB HDDs in RAID 1. And they are great but we put a limit of about 30 concurrent calls through it. That being said we have got larger installs too, we are running 2 of the older 2950's as a fully redundant load balancing pair. For a call center of around 160. The only thing I would watch for is with the 860 the TE110p doesn't work. The TE120p is fantastic no problems but the older card had some incompatibility. Other than that I've never had one skip a beat, so I hope you have the same luck. Cheers, Joel Hill Support Manager Asterisk IT On Mon, 2007-08-27 at 18:15 -0400, Steve Totaro wrote: Steve Totaro wrote: Arthur Miller wrote: Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art **Arthur Miller** Sr. Sales Associate **VoIP Supply, LLC**. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM and a couple 250gig SATA drives. Totally VoIP so I cannot comment on cards or interrupts, but so far it has been flawless. I would like to see how many G729/ULAW conversions it could handle. How would I go about benchmarking that? Thanks, Steve Drooling... processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 1 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 1 siblings: 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 2 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 0 siblings: 2 core id : 1 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 3 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 1 siblings: 2 core id : 1 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address
Re: [asterisk-users] Delete voicemails after X days
Sorry to say I have to disagree with you but I just had a heap of old Voicemails which I couldn't be bothered deleting through my phone, So I went in to /Old/ and ran rm -f on the first 20, I then had to listen to another that wasn't deleted and it was still accessible from the phone, upon further investigation asterisk has renamed them starting again at 0. So running a CRON job to do the same thing should work fine. Cheers, Joel On Tue, 2007-05-22 at 20:37 -0500, Eric ManxPower Wieling wrote: David Florella wrote: Thank you knox. Finally, I have chosen this solution : find /var/spool/asterisk/voicemail/default/*/Old/ -atime -7|xargs rm –f, executed every night by the CRON. However, I would have preferred this feature was implemented in Astrisk. You should expect this to massively break voice mailboxes. Asterisk Voicemail requires that all messages are numbered sequentially starting at 0 (when using the filesystem, I don't know about RealTime or IMAP). If there is a break in the sequence, such as would be the case if your script deletes a message in the middle, then you should expect things to break. I think that higher numbered messages would simply not be accessible, but that is a guess. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vicidial
Hi I'm looking for some help with Vicidial, If you have experience with it and could help with some consulting please contact me off list. Cheers, Joel Hill Asterisk IT [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX and SETLANGUAGE delays
Hi all, I'm having some trouble. I've got an IVR with 4 different languages using the SETLANGUAGE command and I'm getting a 6 second delay when I make the first selection, after that all is fine. There's nothing in my dial plan that I can find that would be causing it. And the delay is driving me nuts! I have an IAX connection from a provider coming in. Could this be the cause? Has anyone experienced anything similar. Thanks, Joel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IAX and SETLANGUAGE delays
Hi Jonathon,
Here's the relevant part (I hope!)
exten = _XX5,1,Answer
exten = _XX5,2,Set(COUNT=0)
exten = _XX5,3,Wait(1)
exten = _XX5,4,Background(ST1000-001-1) ;english
exten = _XX5,5,Background(STS1000-001-1);spanish
exten = _XX5,6,Background(STG1000-001-1);greek
exten = _XX5,7,Background(STI1000-001-1);italian
exten = _XX5,8,WaitExten(1)
exten = 1,1,Set(LANGUAGE()=english); english
exten = 1,2,Goto(STE1050,s,1)
exten = 2,1,Set(LANGUAGE()=spanish);spanish
exten = 2,2,Goto(STE1050,s,1)
exten = 3,1,Set(LANGUAGE()=greek) ;greek
exten = 3,2,Goto(STE1050,s,1)
exten = 4,1,Set(LANGUAGE()=italian);italian
exten = 4,2,Goto(STE1050,s,1)
exten = 7,1,Goto(incoming,_XX5,1)
exten = 8,1,Set(COUNT=$[${COUNT} + 1]) ; after pressing 8 2 times then
goes to consultant
exten = 8,2,GotoIf($[${COUNT} = 3]?4:3)
exten = 8,3,Goto(incoming,_XX5,4)
exten = 8,4,Goto(STE1850,s,1)
exten = 9,1,Playback(STE-thankyou) ; hangs up after plays thank you
for calling msg
exten = 0,1,Goto(STE1850,s,1) ; sends to consultant
Cheers,
Joel
On Wed, 2007-05-02 at 22:49 -0400, Jonathan Barratt wrote:
Hi Joel,
6 seconds sounds suspiciously like Asterisk's dialplan timeout value.
Perhaps you have a wildcard extension that it's waiting to match
against. Either post the relevant section of dial plan or send it to me
off-list, as you prefer, and we'll see what we can find...
Best,
Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Hill
Sent: Wednesday, May 02, 2007 9:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX and SETLANGUAGE delays
Hi all,
I'm having some trouble. I've got an IVR with 4 different languages
using the SETLANGUAGE command and I'm getting a 6 second delay when I
make the first selection, after that all is fine. There's nothing in my
dial plan that I can find that would be causing it. And the delay is
driving me nuts!
I have an IAX connection from a provider coming in. Could this be the
cause? Has anyone experienced anything similar.
Thanks,
Joel.
___
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Re: [asterisk-users] Zaptel kernel module load order
It's generally not recommended to put an analog and digital card in the same box, however that being said Try this. Write a little hack in /etc/rc.local /sbin/modprobe wct4xxp sleep 5 /sbin/modprobe wct4xxp sleep 5 /sbin/ztcfg sleep 5 /sbin/modprobe wctdm sleep 5 /sbin/ztcfg /usr/sbin/safe_asterisk the rc.local script is loaded after all the others so it won't effect anything else, and we had some trouble with some low heat VIA motherboards so we did the modprobe twice for the PRI. Hope this helps. Cheers, Joel Hill Support Engineer Asterisk IT On Mon, 2007-04-30 at 19:14 -0500, Mitch Jackson wrote: Evening, My latest asterisk box is having a difficult problem. It is configured with one TE210P and TDM400P with four FXO modules. I'm running FC6. The TE210P only has a single PRI. When the system boots, it is completely random what order the zaptel modules will get loaded in. Sometimes zttool shows the FXO as the last span, sometimes as the first. When it does load as the first, which happens more often, nothing will initialize properly. When this happens, I have to unload all the zaptel modules, and re-load them over and over again, until the hardware comes up in the correct order. The order it is loaded is in no way related to what order I load the modules on the command line. This problems makes it unlikely that asterisk will start properly if the system is rebooted. Is there something I can do to ensure the modules get loaded in the correct order? Here's my config files, if they will help... # cat /etc/zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us span=2,1,0,esf,b8zs bchan=25-47 dchan=48 defaultzone=us loadzone=us fxoks=49-52 defaultzone=us loadzone=us # cat /etc/asterisk/zapata.conf [channels] language=en switchtype=national context=incoming faxdetect=none signalling=pri_cpe group=1 echocancel=yes resetinterval=never channel = 1-23 language=en switchtype=national context=incoming faxdetect=none signalling=pri_cpe group=3 echocancel=yes resetinterval=never channel = 25-47 signalling=fxo_ks usecallerid=yes callerid=Fidelity Reserves group=2 threewaycalling=no context=outgoing channel = 49-52 Thanks for any help, /Mitch ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Nagios asterisk monitoring
Let me also add my interest, we've got a site using Nagios and haven't had time to work anything out yet related to Asterisk. Cheers, Joel. Joel Hill Support Engineer Asterisk IT On Wed, 2007-04-11 at 18:42 -0400, Watkins, Bradley wrote: Allow me to register my interest in any and all things that tie Asterisk information to Cacti. We use that here, and it's been on my to-do list for a lgg time. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon Kruse Sent: Wednesday, April 11, 2007 6:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nagios asterisk monitoring Yes, I have actually written a resource module for asterisk and the gui to use rrdtool to make REAL pretty gradient shaded graphs based on asterisk data. So, if you want the cacti script, email me([EMAIL PROTECTED]) to get me motivated to rewrite it and make it awesome, and encouragement would be great. But, with a pbx not a pretty graph maker, I am now working on clientside graphing using svg(z) and doing httprequests to get manager information. Let me know if you are interested in that also, I didnt realize how much of a community was out there for monitoring :] -brandon The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with TE110P
Give this a try it fixes a problem we have had with a couple of Via boxes. modprobe wcte11xp modprobe wcte11xp ztfcg -vv zttool We found that probing the card twice before running ztcfg helped alot. Cheers, Joel. Joel Hill Support Engineer Asterisk IT 03 8320 8100 On Mon, 2007-04-02 at 14:56 +1000, Klaverstyn, David C wrote: Type in cat /proc/zaptel/* displays Span 1: ZTDUMMY/1 ZTDUMMY/1 1 But if I type in lsmod | grep -i wct I get wcte11xp 26016 0 wct4xxp 221120 0 zaptel184996 3 ztdummy,wcte11xp,wct4xxp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, 2 April 2007 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problems with TE110P On Mon, Apr 02, 2007 at 12:50:18PM +1000, Klaverstyn, David C wrote: I have a new server using Zaptel 1.2.16 Issuing a ztcfg gives the following error: ZT_CHANCONFIG failed on channel 1: No such device or address (6) loadzone=au defaultzone=au span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17-31 dchan=16 Now what do you actually have loaded? cat /proc/zaptel/* ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configurations Files of TE110P
Here you go. This is from ATP http://www.austechpartnerships.com/forum/viewtopic.php?t=76 /etc/zaptel.conf loadzone=au defaultzone=au span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 dchan=16 This assumes connection to a carrier, where they provide clocking. /etc/asterisk/zapata.conf switchtype = euroisdn signalling = pri_cpe group = 1 pridialplan=unknown context = incoming channel = 1-15 channel = 17-31 Cheers, Joel On Sun, 2007-03-04 at 23:41 +, younss azzayani wrote: please can someone send to me his files like zaptel zapta if he si using TE110P thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dell 860
Hi All, I'm having some troubles with my Dell 860 and TE110P card. Using Asterisk 1.2.14, Zaptel 1.2.12 and Libpri 1.2.4. I'm getting digital noise, like a half ring almost and other jitter. Here's the kicker it's only on the outside part of the call. Ie. if I rang you, you would here it but I don't and the opposite if you rang me you would hear it but I wouldn't. I have tried two different cards and different E1 lines still the same thing? I'm going to try the two port card soon but I don't think that will fix my problem. Is it just one dodgy server or are all 860's no good? Thanks for your help. Joel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compatability
Hi All, I have a new client who has an existing Asterisk PABX and is looking for us to install a TE110P for him, However he has a Dell SC420 and I have never used one before. I have had no problems with any other Dell servers which we use almost exclusively. Has anyone had any good/bad experiences with the SC420 in relation with Digium cards? Thanks for your help. Joel Asterisk IT www.asteriskit.com.au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snom 360: how to make record button working ?
Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] snom 360: how to make record button working ?
No worries. Good question, I wasn't sure so I just tested it and it seems that the answer is yes it does send the tones to the other side. Can I ask why this would matter, I think there could be legal implications of recording a call and not notifying the other party. That's why you always get the message This call may be monitored for training and coaching purposes. Etc.. Cheers, Joel. Remco Barendse wrote: Thanks for this, I was looking for this too. Will the DTMF tone be audible to the other side? (In other words will they know something is happening) On Thu, 5 Oct 2006, Joel Hill wrote: Hi Noro, Depending on what firmware you have this is the way to go. Go to the Functions keys page, then look for the Record button, Change the type to DTMF and in number put in *1 which is the default Asterisk recording function. Hope this helps Cheers, Joel Asterisk IT www.asteriskit.com.au noro kamen wrote: Hi, I'd like to make record button working on snom 320/360 + asterisk. As I learned from wireshark output, the phone produces SIP info message Record: on, while record button pressed. Can anybody give me an advice, how to teach asterisk to understand that SIP info message and start recording ? TIA noro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sound Quality.
Hi I'm getting really bad static on forwarded calls to the point of not being able to hear the person at the other end. I'm running an E1 line in and everything else is fine. I'm also getting this error: Sep 11 14:56:56 WARNING[18295]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/64) I'm running Asterisk 1.2.11 and have tried a couple of different codecs in SIP.conf. Any ideas?? Cheers, Joel Hill Support Engineer Asterisk IT www.asteriskit.com.au www.theasteriskshop.com.au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
