[asterisk-users] Poor-man's paging through multiple phones?
We're mostly Cisco CallManager with some SIP and Asterisk. I want someone at one of our locations to be able to dial and number and have Asterisk simultaneously dial several Call-Manager extensions which are set to auto-answer and talk into the phone creating a sort of paging system. We have informacast, but it is too cumbersome for the users. I know Asterisk can ring several phones at the same time... if one of them answers, the others stop right? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer BACK to CallManager over SIP trunk?
We have occasional problems with failed transfers. The PSTN call comes into Cisco Call Manager, then to asterisk over a SIP trunk and then to an asterisk controlled SIP phone. The SIP phone transfers back to a CallManager controlled SCCP phone which sometimes fails. Is there a wait to let CallManager handle the transfer instead of asterisk? I have a feeling asterisk is handling the traffic even after the call is transferred. ___ -- 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] To what degree can Asterisk replace Cisco Unity?
In a CallManager environment (currently 4.0, moving to 6.1 in the next few months), can Asterisk completely replace Unity as a voicemail system? What works and what doesn't? MWI? Call Handlers? Does everything work via a SIP trunk? Who has done this and is willing to contact me? Thanks. ___ -- 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] CCM 6 and Asterisk routing again
I've noticed two differences in what you described and my working CM setup: 1. My sip trunk in CM is defined as 711alaw, you have ulaw. 2. My sip.conf defines CM as a type=friend instead of a peer. Do you have any SIP phones connected to Asterisk (you could use a softphone like the free xten)? Can you call the phone from CallManager? Peter Pauly http://www.usbtests.com On 3/11/08, Aaron Fransen [EMAIL PROTECTED] wrote: Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1, Asterisk is running strictly VoIP over the network and using CCM as the trunk. Calls from the SIP phones connected to Asterisk work fine. They can call both external numbers and any Cisco extensions attached to CCM. Calls from CCM to Asterisk fail without any notification in Asterisk (and I DID have this working at one point, but I suspect that our Telco may have pooched the config somehow, since they're in the process of connecting us to another CCM site). I have verified: Media Termination point exists, Calling Search Space exists, Trunk is properly defined (uLaw 711, UDP, ip address port, etc), and a route pattern exists to take calls to the right trunk. The system will let me complete the dialing sequence to the Asterisk server, but as soon as I enter the last digit I get a busy signal. Thoughts anyone? Here's my sip.conf if that helps... [callman] type=peer context=incoming insecure=very host=(ip of my call manager server) disallow=all allow=ulaw allow=alaw nat=no canreinvite=yes qualify=yes Thanks! Aaron ___ -- 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] Any phone capable of displaying real time queuestatistics?
I don't see any evidence that queue metrics can push data to the phone. I'm really looking for a home-grown solution that pushes XML/HTML to a phone during a call, like the 7960's. On 12/13/07, Dovid B [EMAIL PROTECTED] wrote: Queue Metrics - Original Message - From: Peter Pauly [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, December 11, 2007 9:06 PM Subject: [asterisk-users] Any phone capable of displaying real time queuestatistics? Are there any phones whose display can show queue statistics, ie: calls waiting, etc, on the phone itself without too much trouble with Asterisk? Especially while the phone is in use (on a call)? ___ --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 ___ --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] CallManager sip trunk - callerid name?
I have been unable to get callerid name passed from Cisco Callmanager over a SIP trunk to Asterisk. Only the number is displayed. Has anyone been successful getting callerid name? ___ --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] Any phone capable of displaying real time queue statistics?
Are there any phones whose display can show queue statistics, ie: calls waiting, etc, on the phone itself without too much trouble with Asterisk? Especially while the phone is in use (on a call)? ___ --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] SIP 7960 soft key customization?
Does anyone know how to customize the order of the soft keys on a 7960 running SIP? All the documentation I could find is CallManager related. Specifically, I want to move the transfer function to the first set of buttons during a call. ___ --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] Suppressing certain queue announcement voice prompts
Try this: queue-thankyou = /dev/null On Nov 30, 2007 10:02 AM, [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Short of replacing a sound file with a sound file containing only a short period of silence, is there any way to suppress certain sounds from playing during queue processing by configuring for example queues.conf or other similar files? Which announcements are you trying to not play? queue-thankyou for instance, to name one. Or any other of the queue-* files in general. From time to time it can be convenient to change the exact prompts played (order and contents) due to language differences and personal preference of the end-users. We're doing this now by replacing them with silence but I'm just thinking that it would be more elegant to have Asterisk not attempt to play them in the first place. We've also removed the files in some instances but that's even worse from my point of view because then we get file-not-present warnings. ___ --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] Don't enter a queue if no one is logged in
I currently have the following setup: exten = 2000,1,Playback(/var/lib/asterisk/sounds/Greeting) exten = 2000,2,Queue(Qabcdef|t) exten = 2000,3,Playback(/var/lib/asterisk/sounds/EveryonesBusy) exten = 2000,4,Hangup exten = 2000,103,Hangup What happens is, that the greeting in step one is played regardless if anyone is logged into the queue. So immediately after the greet, we tell them we can't help them. What I would like is to check first to see if there is anyone logged into the queue, and then play the greeting. Is this possible? Is there a function that checks if anyone is logged in? ___ --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] Cisco 7960 won't download dialplan.xml
I'm monitoring my tftp servers' logs and my Cisco 7960 test phone won't download dialplan.xml to the phone. I know this from the logs and from the behavior of the phone. I see it downloading other files like the ring tone file, etc. Is there something that needs to be set in the cnf files to download the dialplan? I thought it is included automatically. I've also tried reseting the phone to factory presets. I'm running POS3-08-2-00. ___ --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] Cisco 7960 vs 7905
The screen on the 7960 is a rather low resolution one, and therefore does not display much data. Pressing the directory button (and selecting Resolutions and color depth on the phones are as follows: 7905/7912 192x53 Grayscale, Depth=1 7920 128x59 Grayscale, Depth=1 7940/7960 133x65 Grayscale, Depth=2 7970 298x168 Color, Depth=12 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Phones - Power over ethernet?
Are there any cheap SIP phones (like the Grandstream for example) that support power over ethernet? What is necessary to support SIP phones in a Cisco Call Manager environment? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: SIP vs. Skinny protocol
I assume there are several people on this list that have Cisco Call Manager implementations under their belt We are beginning a call manager implementation and the first question I asked Cisco was, should we use SIP or Skinny. Cisco is pushing me towards Skinny, saying that I will lose some functionality with SIP. They also say that most of their customers implement skinny. I see two obvious benefits to using SIP: 1. I can get cheaper phones that run SIP, altough Cisco just came out with a 7902G for $130 US. 2. It's an open protocol and is more likely to survive long-term. What functionality do I lose by going with Skinny? Will Cisco eventually go with SIP only and I'll have to convert anyway? Any other pluses or minuses? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk in a Centrex environment?
Does anyone know what would be involved in making Asterisk work as a voicemail system in a Centrex environment? We have a Centrigram voicemail system that belongs in the Smithsonian. There are analog lines coming into the box and a 56KB data feed from the phone company's switch. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Seeking proposals for large county library voice system
The Indianapolis Marion County Public Library has put out a request for proposals on its website for a local dialtone/voice system. I know there are several people on this list that run consulting companies that specialize in implementing Asterisk systems. A PDF file describing the RFP process can be found at: http://www.imcpl.org/rfp03_local_voice_service.pdf I would also appreciate any information about large scale implementations of Asterisk. FYI: I have been a member of this mailing list for several months and run Asterisk in a test environment. Peter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Consultants/Companies in Indianapolis?
Are there any companies/consultants in the Indy area that are Asterisk experts? Please contact me via email. THanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] initial review of Grandstream HT-286 ATA device
The PDF on the website says that this thing supports a downloadable ring-tone. This makes me somewhat suspicious - does this thing generate ringing voltages and actually ring the attached analog phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware?
I'm currently running firmware version 3.2 on my Cisco 7960. I've seen on the list that several people are running the 5.x latest versions. I've avoided going to higher firmware versions because I'm worried about potential problems or issues with the encryption mechanism used in the later firmware versions. (Once you go to an encrypted firmware version, you can't go back, right?) For those of you who have gone to the newer firmware, what features or benefits have you seen by going with the newer firmware? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Cisco Color Phone
I thought you guys would be interested to know: eWeek has a short article about Cisco bringing out a new IP phone: 7970G. It has a high resolution color touch-screen display with support for XML and can act as a mini-browser to allow the development of vertical applications. But get this: the price will be $995. I don't think I'll be getting one any time soon. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Radio for Music on Hold?
On Thu, Sep 18, 2003 at 01:21:54PM -0700, Paul Crick wrote: Come on people! Fork out $50 for a discman and another few bucks for some royalty free library music and have that on hold instead.. You're in control, you know what your callers are listening to, and you're also legal Why go to all this trouble and expense? - skip the Discman and just rip the royaltyfree CD and save the mp3's on the hard drive. (Check the license to make sure you are allowed to do this). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoicePulse offering IAX2 services
I don't know if this has been mentioned yet: Voicepulse is now offering wholesale pricing and IAX2 connectivity for Asterisk users. No fees, pay as you go. They also offer incoming calls for $7.99 per month. See wholesale.voicepulse.com. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone 800 numbers working?
On Thu, Sep 18, 2003 at 07:02:42AM -0700, TC wrote: well, i have same problem... it sounds like nufone is not allowing calling of #800. anyone from nufone care to comment? I have seen nufone die, if the callerid is not a cid from us 48 try setting your sic to I added SetCallerID and SetCIDName steps before the dial and it works now. Funny, it worked before without these steps. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nufone 800 numbers working?
Is anyone else having trouble dialing 800 numbers through Nufone? I'm getting the SIT tones no matter what number I dial. Normal long distance works fine. I don't think it's my dial plan (it was working previously). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Source for 50-pin amphenol cables?
I'm looking for a source for 50-pin amphenol cables, the ones used to connect Adtran's to punch down blocks. Preferably, one that's mail order and takes orders over the internet. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Start of all recordings cut off
On Thu, Sep 11, 2003 at 09:30:35PM -0700, John Todd wrote: Before running any application that has sound playback (Playback, Background, VoiceMailMain2, etc.) it would be wise to execute an Answer first, then a Wait(2) to allow for VoIP channels to fully establish and settle. Adding Answer had no effect. Adding Wait(1) solved the problem. Maybe it's because Asterisk runs on a slow machine (750Mhz P3). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phpconfig is out in CVS
On Thu, Sep 11, 2003 at 10:12:50PM -0600, Dave Packham wrote: nope when I click on something on the left I get a FQDN not just the pne you had Hmmm. Further info: it works with Microsoft Internet Explorer. It does not work with Mozilla 1.4 under Linux. It also does work with Mozilla Firebird under Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Start of all recordings cut off
I'm using a Cisco 7960 with asterisk and any recording on the machine, be it voicemail prompts, time of day, echo test message, etc, is cut off for the first 1/4 to 1/2 second. I've tried setting the phone to gsm but it still happens. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phpconfig is out in CVS
On Thu, Sep 11, 2003 at 08:42:18PM -0600, Dave Packham wrote: hmm works for me... its the exact same code that is installed on the sample server listed below and I dont get the problem there. lemme know more info and ill look into it Dave Well, there is no such domain as phpconfig. It's probably pulling this file off of your machine. However, you've suceeded in teasing us - it looks very cool so far. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets
On Tue, Sep 09, 2003 at 02:38:01PM -0500, Eric Wieling wrote: That would be reinvite= and canreinvite= in the user entry for each SIP endpoint. Asterisk will allow the endpoints to talk directly to each other if both those settings are = yes (the default, I think) AND both endpoints use the same protocol (SIP) AND the same codec. This is the single most useful bit of info I have seen on the mailing list since I have joined. Thanks Mr. Wieling. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xlite = no sound
On Tue, Sep 09, 2003 at 11:41:17AM +0100, Skuse, Phil wrote: Yes. They are on the same subnet. I solved my sound problem with X-lite by using the latest CVS version and compiling that. I had been using the stable and unstable versions out of Debian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xlite = no sound
On Tue, Sep 09, 2003 at 11:04:34AM +, WipeOut . wrote: Where did you get access to X-Ten.com's CVS server? I didn't know they had the source code for x-lite available.. Sorry, I should have been more clear - I used the latest version of Asterisk via CVS. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mixed FXO and FXS on one Adtran, T1 card?
Can I configure an Adtran channel bank with a mixture of FXS and FXO cards and have them come into a single T100P T1 card? It seems like this would be a cheaper solution than trying to load a bunch of PCI cards into a PC. Also, when shopping for an Adtran (on ebay) - what do I need to watch out for as far as interfaces or capabilities to make sure that it works with a Digium card? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still no audio on SIP phone
For the benefit of others having this problem - I installed the latested CVS build and the problem went away - I can hear audio now from X-lite. I was using the debian unstable package. Here's what I did: cd /usr/src mkdir asterisk export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login (supply the anoncvs password here when prompted) cvs checkout asterisk cd asterisk make make install make samples make sounds (I think that's right - memory's getting fuzzy from age) added my extention back into sip.conf works like a champ. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modems and Tivos? Oh my!
Does the Digium FXS card support modems (and Tivo devices)? If so, to what speed have they been tested? Also, on a somewhat unrelated question: How does the FXS card generate ringing voltages if the PC only supplies 12 volts? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modems and Tivos? Oh my!
On Thu, Sep 04, 2003 at 06:26:03PM -0500, James Sharp wrote: On Thu, 2003-09-04 at 17:22, Peter Pauly wrote: Does the Digium FXS card support modems (and Tivo devices)? If so, to what speed have they been tested? Assuming that you can do native zaptel bridging (Going from an FXS port to an FXO port in the same machine), you should be able to get up to 33.6. No 56k, unfortunately, because of the multiple D/A A/D conversions. If you're codecing the audio and passing it over IP, you should be able to get 33.6 if you use ulaw (non-compressed) encoding. Any of the compression based codecs will most likely make your modem link up at 9600 or a flaky 14.4. Why would one use dialup for a TiVo. My TiVo has never touched a telephone line ever. When I bought it I hacked it to work over my cable modem link using PPP to my workstation and have never risked loosing the internal hardware. Or spend a few bucks and get yourself one of the ethernet card kits. Then you don't have to worry about ppp connections and you can drag the video off the unit as well. I can't use the networking kits because it is a DirecTivo HDVR2. Supplying this thing with dial-tone is a consideration for me before I dump my POTS line. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still no audio on SIP phone
adding nat=yes to the sip definition made no difference. Does Asterisk use the DSP in your sound card to do the audio processing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem solved - sort of..
I started to suspect the X-lite client in my problem (I was getting no audio when calling into asterisk) because after I would make test calls to asterisk, setting X-lite back to my FWD account - I would get no audio with FWD either, even though the sound card was working and I got dial-tone, etc. I tried the SJPHONE client - and low and behold - asterisk works fine. So it's definately not an asterisk problem. Something flaky with X-lite. Peter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Still no audio on SIP phone
I have been using X-Lite on FWD without any troubles and recently became interested in trying asterisk. I am able to register from X-Lite and dial a number - I've tried dialing some of the sample numbers in the sample extentions.conf file, like 500 and 1234, they appear to dial correctly from X-lite but nothing else happens - no audio is heard. My understanding is that I should hear some sort of message. I already found one problem - on my debian system - /usr/bin/mpg123 was a symbolic link pointing to mpg321. I've corrected that and installed mpg123/unstable and made sure it was the real deal (deleted the symbolic link, etc). I am still not getting any audio. My setup: Debian (from Knoppix - a mix of unstable, testing, stable), no hardware phone cards, one software SIP phone (X-lite). Everything is on a LAN (no firewall involved). Where should I begin to find this problem? I've tried starting asterisk with lots of verbose flags, but I don't see anything suspicious. Peter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still no audio on SIP phone
On Tue, Sep 02, 2003 at 03:28:11PM -0600, Gavin Hollinger wrote: correctly from X-lite but nothing else happens - no audio is heard. My understanding is that I should hear some sort of I am using x-lite with the asterisk demo no problem. All I modified was sip.conf Is the asterisk server and your x-lite client on the same LAN segment? Is all iptables and firewall code turned off on the asterisk server? Gavin Hollinger Here is the message I am getting from Asterisk: *CLI -- Executing VoiceMail(SIP/2000-3296, u1234) in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm/1234/unavail' -- Playing 'vm-intro' -- Playing 'beep' -- Recording to /var/spool/asterisk/vm/1234/INBOX/msg0001 WARNING[229391]: File app_voicemail.c, Line 673 (leave_voicemail): No audio available on SIP/2000-3296?? -- User hung up It shows it is playing the files, but nothing is heard on the Xlite SIP software side. When Asterisk starts up, it complains about OSS and ALSA problems - sound capabilities on the console are irrelavent in this case aren't they? I've tried deactivating several different codecs in X-lite - it doesn't help. Peter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
