[asterisk-users] FastAGI: how does remote process learn of hangup/disconnect
Hello! I m reading all sorts of descriptions of FastAGI() on the internet but I m finding no information how asterisk signals the remote process that the connection has been hungup. In an AGI script you receive a SIGHUP. What happens with a FastAGI connection? Regards ___ -- 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] [Softphones] ZoIPer vs. XLite?
Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively. For those of you have tried both, especially in a context with NAT firewalls on both sides, what's the outcome? Did you stick to ZoIPer, or is it not good on par with X-Lite? Are other clients I should know about? http://www.zoiper.com/ http://www.counterpath.com/ 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
[asterisk-users] Can't dial out from SIP to CAPI
Hi, I've been trying to configure my extensions.conf and sip.conf for two days now and I'm pretty sure it's just a small typo or anything I can't find by myself. My setup: - Asterisk connected via Fritz! PCI Card to a HiPath 3500 (2 channels) - Callcentric.com SIP channel to dial out to foreign countries - Cisco 7912 attached to asterisk using SIP (in another city) When I dial extension 85 my Cisco phone is ringing and I can talk and everything works fine. But when I try to dial an extension from the dialplan I never get a connection. I've posted my capi.conf, extensions.conf and sip.conf here: http://pastebin.com/f19940490 When dialing out via ISDN I have to dial a 0 to get a line. Something I notices in the SIP debug is, that my phone always requests numbers like sip:[EMAIL PROTECTED];user... , when dialing out. Is that right when using a SIP device? I have not tried the callcentric stuff so far and that's not so important for me right now. I just want to be able to dial out in the first step. Maybe you find my problem. Thanks, Sebastian. ___ -- 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] Got SUBSCRIBE for extension ... but there is no hint for that extension.
Hello, This is just a warning, that a snom phone tries to subscribe an extension which has no hint entry. You should try to find the snom which has set up the subscription which wasn´t found. You should search in the snom webif in the function keys for the function nebenstelle or extension in english. Best regards. Stefan Stefan Guenther schrieb: Hi, in the CLI I get a number of messages telling me Got SUBSCRIBE for extension ... but there is no hint for that extension. The call is established anyway, but what causes these messages? Obviously it is a problem/feature of SNOM phones because every other phone that we use, doesn't produce these messages. I would be happy if someone could tell me what do to to get rid of these messages. BTW, I use subscriptions to monitor other extensions. Thanks for your help. Stefan ___ -- 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] Mistake in the wiki's description of cmd Pickup() ?
Hi, according to the description of Pickup() on page http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup I can use this command to pickup a call at a certain extensions. When I try this with e.g. exten = *8200,1,Pickup(200) Asterisk tells me that the highest value for the Pickup command is 63. Wenn I enter the number of a callgroup instead of an extension, I can pickup the call. Well, do I misunderstand something or has the behaviour of Pickup() changed? I have tested this with Asterisk 1.4.17-BRIstuffed-0.4.0-test6 Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- 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] Zaptel timer on Intel Dual Core servers
Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Feb 02, 2008 at 09:46:42AM +0100, Johansson Olle E wrote: Friends, I'm having severe problems with zaptel timers on Intel Dual Core systems with SMP code enabled. Ztdummy, zaptel connected to Digium TDM or PRI cards - all ends up with large timer probems - zttest going down to 50% accuracy on some systems, even to -1 on ztdummy systems and voice quality is no more. A restart is the only way to get back to a working system. A restart of asterisk? Not touhcing the kernel modules? What CPU is it, exactly? /proc/cpuinfo would help. What version of Zaptel? Asterisk? Kernel? distribution? Olle, Specifically, what kernel version? If you are using 2.6.9 (e.g. Centos 4), then by default when you build ztdummy, it does NOT use USE_RTC (although IMHO it should do). You should make the following change to ztdummy.c: #if LINUX_VERSION_CODE = VERSION_CODE(2,6,13) #define USE_RTC #else -#if 0 +#if 1 #define USE_RTC #endif #endif And then recompile. This will make ztdummy use the RTC instead of kernel jiffies. If that's not it, then I hope you discover the cause, and would be interested to know what it is. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- 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] Zaptel timer on Intel Dual Core servers
On Tue, Feb 05, 2008 at 09:58:54AM +, Tony Mountifield wrote: Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Feb 02, 2008 at 09:46:42AM +0100, Johansson Olle E wrote: Friends, I'm having severe problems with zaptel timers on Intel Dual Core systems with SMP code enabled. Ztdummy, zaptel connected to Digium TDM or PRI cards - all ends up with large timer probems - zttest going down to 50% accuracy on some systems, even to -1 on ztdummy systems and voice quality is no more. A restart is the only way to get back to a working system. A restart of asterisk? Not touhcing the kernel modules? What CPU is it, exactly? /proc/cpuinfo would help. What version of Zaptel? Asterisk? Kernel? distribution? Olle, Specifically, what kernel version? As I understand from an IRC chat - CentOS 5 - 2.6.18-something. If you are using 2.6.9 (e.g. Centos 4), then by default when you build ztdummy, it does NOT use USE_RTC (although IMHO it should do). You should make the following change to ztdummy.c: #if LINUX_VERSION_CODE = VERSION_CODE(2,6,13) On a kernel version that calls itself 2.6.9, that part is never reached. So that point is mute. #define USE_RTC #else -#if 0 +#if 1 #define USE_RTC #endif #endif -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] Can't dial out from SIP to CAPI
On Tue, 5 Feb 2008, Sebastian Pape wrote:
Hi,
I've been trying to configure my extensions.conf and sip.conf for two days
now and I'm pretty sure it's just a small typo or anything I can't find by
myself.
My setup:
- Asterisk connected via Fritz! PCI Card to a HiPath 3500 (2 channels)
- Callcentric.com SIP channel to dial out to foreign countries
- Cisco 7912 attached to asterisk using SIP (in another city)
When I dial extension 85 my Cisco phone is ringing and I can talk and
everything works fine. But when I try to dial an extension from the dialplan
I never get a connection.
I've posted my capi.conf, extensions.conf and sip.conf here:
http://pastebin.com/f19940490
Your Dial() line for CAPI:
Dial(isdn/g1/@13:${EXTEN},30,r)
is not correct.
I assume you use a newer chan_capi, then it should look like:
Dial(CAPI/g1/13:${EXTEN}/b,30)
Armin
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Re: [asterisk-users] [Softphones] ZoIPer vs. XLite?
On Tue, 5 Feb 2008, Vincent wrote: Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively. For those of you have tried both, especially in a context with NAT firewalls on both sides, what's the outcome? Did you stick to ZoIPer, or is it not good on par with X-Lite? Are other clients I should know about? http://www.zoiper.com/ http://www.counterpath.com/ Zoiper just works in IAX mode. The GUI is simple and easy to use too. (Although I've only first-hand experience of the Linux one) Gordon ___ -- 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] Losing CALLERID{dnid}
Sorry,
I tried to use underscore(s) before the variable names, but without any
success.
H234m_gw is a functionality which we use for video calling on asterisk.
(http://sip.fontventa.com/)
--
Arjan Kroon
Mobillion BV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared
Smith
Sent: dinsdag 5 februari 2008 1:31
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Losing CALLERID{dnid}
On Mon, 2008-02-04 at 10:08 +0100, Arjan Kroon | Mobillion wrote:
When I setup the videocall with exten =
n,1,h324m_gw([EMAIL PROTECTED]), I loose the variable DNID
(${CALLERID(dnid)})
Hmmmn... I'm not familiar with the h324m_gw application. Is that some
third-party add-on to Asterisk?
Have you tried doing something like:
exten = blah,1,Set(__MY_DNID=${CALLERID(dnid)})
exten = blah,n,h324m_gw([EMAIL PROTECTED])
and see if that MY_DNID channel variable is still set after the call?
(The underscores on the beginning of the variable tell Asterisk that any
child channels should inherit the channel variable from this channel.)
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] [Softphones] ZoIPer vs. XLite?
Jared was talking about a decent IAX hardphone on this list a week or so back, I don't recall the make. If you use IAX, all you need to do is : 1) set your local firewall to forward udp 4569 to asterisk. (optionally filtering by from IP address if your user has a fixed IP address or known range) 2) have your ZOIPer or other IAX phone register with asterisk every 60 seconds. 3) configure an IAX 'friend' account for your user. You should not need to make _any_ changes to the firewall at the remote end (unless they block all outgoing UDP). Tim. - Original Message - From: Vincent [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: 05 February 2008 08:07:01 o'clock (GMT) Europe/London Subject: [asterisk-users] [Softphones] ZoIPer vs. XLite? Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively. For those of you have tried both, especially in a context with NAT firewalls on both sides, what's the outcome? Did you stick to ZoIPer, or is it not good on par with X-Lite? Are other clients I should know about? http://www.zoiper.com/ http://www.counterpath.com/ 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 ___ -- 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] one CDR instead of multiple CDR
This is a part of our programma. [begin] exten = s,1, h324m_gw([EMAIL PROTECTED]) [video] exten = s,1,h324m_gw_answer() exten = s,2,Wait(3) exten = s,3,Goto(intro,s,1) [intro] exten = s,1,mp4play(intro.3gp) exten = #,n,Goto(einde,s,1) [einde] exten = s,n, Hangup() When I use this dialplan and during the intro.3gp I press the #-key the call will be ended. But I got three different CDR's. Does anybody know how I can use one CDR instead of 3 different CDR's Kind Regards, Arjan Kroon Mobillion BV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: maandag 4 februari 2008 15:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] one CDR instead of multiple CDR On 2/4/08, Arjan Kroon | Mobillion [EMAIL PROTECTED] wrote: Hi, In my application I jump to different extensions For example: [begin] exten = s,1,Goto(starts,s,1) [start] exten = s,1,Play(welkom) . exten = h,1,Goto(end,s,1) [end] exten = s,1,Macro(end_call) exten = s,n, Hangup When I look at my CDR record I see three different CDR's in my record. Is there a way to use one CDR on every call, and not multiple CDR on every call? You should post also the relevant sections of your dialplan that manipulates CDR's. For example calls to Dial() or Queue() applications. Also a log snippets (uncomment the full line in logger.conf) that says anything about posting CDR and previous few commands would be useful. Regards, Atis Kind Regards, ___ -- 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 -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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] SIP Channels
Hi, try use Dial with G parameter, and bridge these to extensions with meetme. But the problem is that, I don't know how to close conference, when one hangups... 2007/11/2, Asterisk [EMAIL PROTECTED]: Hi there, I'm trying to bridge 2 SIP channels together via AGI script. The AGI script is written in C#. The first caller would call in and be placed on hold and the second caller would call in and both the calls gets connected together. But I am having problem with the second caller finding the first channel. Can someone point me to the right direction? thanks Eric ___ --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 -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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] app_valetparking.c anyone using it on 1.4?
Hi List, I have this running, but after I park a call it will not announce where it is at, it's like you have to call another application just to say where it is parked at. I have tried a second priority option for the same extension with that ValetParkList but it seems once ValetParkCall has been ended it will not process anymore priorities in this extension. Any ideals or help would be great! I'm using ValetParking with 1.4. ValetParking doesn't announce anything because the whole point of ValetParking is to be able to explicitly park a call at a known spot. We use it to park a call at a users extension when they aren't at their desk, or are on another call. When they're finished with the call or they're paged all they have to do is dial their known park location. I haven't had any problems with priority options. -- Marvin Horst ___ -- 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] wireless VOIP phone recommendations?
I have been using the D-Link DPH-540 wireless VOIP handset, and I really like this phone. We had tried the UStarcomm phone, but the phone is used in a noisy environment and the volume wasn't loud enough. The problem with the D-Link phone is the Li-ion battery needs to be replaced and D-Link doesn't sell a replacement battery and I haven't found any after-market batteries. So this phone is essentially a brick because I need a new battery :( So any recommendations for another wireless VOIP phone? -- Marvin Horst ___ -- 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] one CDR instead of multiple CDR
On 2/5/08, Arjan Kroon | Mobillion [EMAIL PROTECTED] wrote: This is a part of our programma. [begin] exten = s,1, h324m_gw([EMAIL PROTECTED]) [video] exten = s,1,h324m_gw_answer() exten = s,2,Wait(3) exten = s,3,Goto(intro,s,1) [intro] exten = s,1,mp4play(intro.3gp) exten = #,n,Goto(einde,s,1) [einde] exten = s,n, Hangup() Seems like a side-effect of using local channels. You could try adding NoCDR() in context video, and see if that helps, and you still get valid call durations. Or as alternative - add NoCDR in context begin, as it completes almost immediately. However i don't see where third channel is raised.. Could you provide debug logs of affected call from /var/log/asterisk/full (enabling full in logger.conf). Regards, Atis When I use this dialplan and during the intro.3gp I press the #-key the call will be ended. But I got three different CDR's. Does anybody know how I can use one CDR instead of 3 different CDR's Kind Regards, Arjan Kroon Mobillion BV -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: maandag 4 februari 2008 15:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] one CDR instead of multiple CDR On 2/4/08, Arjan Kroon | Mobillion [EMAIL PROTECTED] wrote: Hi, In my application I jump to different extensions For example: [begin] exten = s,1,Goto(starts,s,1) [start] exten = s,1,Play(welkom) . exten = h,1,Goto(end,s,1) [end] exten = s,1,Macro(end_call) exten = s,n, Hangup When I look at my CDR record I see three different CDR's in my record. Is there a way to use one CDR on every call, and not multiple CDR on every call? You should post also the relevant sections of your dialplan that manipulates CDR's. For example calls to Dial() or Queue() applications. Also a log snippets (uncomment the full line in logger.conf) that says anything about posting CDR and previous few commands would be useful. Regards, Atis Kind Regards, ___ -- 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 -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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 -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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] [Softphones] ZoIPer vs. XLite?
Marc Charbonneau wrote: - shameless plugMy MediaX softphone : http://www.marccharbonneau.com/asterisk/mediaxphone.php/shameless plug Marc, does your client play nicely with Vista? We've been having some problems with softphones that work fine in XP, but choke in Vista. -- Alan Williamson Professional Self Publishing Packages http://www.Blog-City.com/ myBlog = 'http://alan.blog-city.com/'; ___ -- 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] [Softphones] ZoIPer vs. XLite?
Are other clients I should know about? http://www.zoiper.com/ http://www.counterpath.com/ Add to that list - Mozphone (http://mozphone.mozdev.org/) that can be installed in Firefox -Kiax : http://sourceforge.net/projects/kiax - shameless plugMy MediaX softphone : http://www.marccharbonneau.com/asterisk/mediaxphone.php/shameless plug - iaxcomm : http://iaxclient.sourceforge.net/iaxcomm/ - The one from Sokol associates : http://www.sokol-associates.com/?q=node/29 hth ___ -- 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] wireless VOIP phone recommendations?
We're using Pirelli DPL10's and nokia N95's with cisco aironet access points and both phones are quite happy roaming around the building (6 access points) during calls - the nokias seem to have better signal strength and audio quality than the pirelli's though. Geraint SIP wrote: Brian J. Murrell wrote: On Tue, 2008-02-05 at 14:37 +, [EMAIL PROTECTED] wrote: Hi, I use Linksys WIP 330 and the sound is good, talk time with full battery go up to 2 hours, I'm happy with. Ahhh. OP wanted to know about wirelessly networked phones. Interesting as they are (and expensive -- the WIP-330 retails for $229 at voiplink.com), I was hoping this would be a thread about simply cordless IP (SIP or IAX) phones. I think these tend to be available at a more reasonable price. I have a Panasonic GLOBALRANGE BB-GT1500CB (http://www.panasonic.ca/english/telecom/telephones/globarange/index.asp) which technically is supposed to only work with the Joip service, but spoofing this phone to work with your Asterisk server is not too difficult. It's a reasonable phone at a reasonable price (CAN$70 the last time I looked at a retail shop) but I have found that it can drop out sometimes. I tend to think the drop-out is in the audio handling in the handset itself rather than anything on the network. It seems like it might be some kind of silence detection and optimization circuitry (i.e. not transmitting dead air to the base station) that just doesn't work in real-life as well as it did on paper. Also this Panasonic phone does not do call-waiting. When there is a call in session on it, an attempt to route a second call to it from Asterisk results in a busy here message back from the phone. :-( I wonder what else is out there in a more affordable consumer price range. I guess there is always ATAs and regular phones. I've always wondered though if there is any benefit to even a basic phone such as the GLOBALRANGE phones being native SIP vs. just using an ATA. I have not discovered anything this phone can do above and beyond what our standard cordless Panasonic phone does plugged into an ATA. b. ___ -- 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 For the most part, for day to day dialing, you won't see any really significant difference between a WiFi phone and an ATA with a regular cordless or DECT phone. You may lose the ability to dial SIP URIs (although not all wifi sip handsets have this ability). However, in general, none of the true Wi-Fi phones we've tested other than the Nokia E series have been worth mucking with. Dropping off APs, poor NAT capability, low battery life, troublesome configurations, random weirdness -- these seem to abound in the world of wi-fi SIP. This is why the usual scenario for any sort of office-wide deployment involves DECT. It's a shame, really. With wi-fi being so prevalent so many places we go, and with the possibility for portability being outstanding, it's a shame the hardware manufacturers haven't quite made anything worth buying. N. ___ -- 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] is encrypted iax safe and secure?
Hello, I'm doing some research concerning iax encryption, I haven't find any clients (softphones or hardphones) which implement so I have not tested it yet. There was also this message on asterisk-security mailing list http://archives.free.net.ph/message/20070507.101933.222987b2.en.html which got no answers and this makes me think that this iax encryption is not much interesting for the community. Anyway, in iax specification there is this statement: Only the data portion of the messages are encoded. Which are the consequences of this, is it true as stated on http://www.voip-info.org/wiki/view/IAX+encryption that The calling/called numbers are still passed in the clear over encrypted IAX, so you are still vulnerable to traffic analysis. ? If it's true how to deal with this? Would you consider media payload encryption enough? Maybe it's better to just forget about iax encryption and consider some more general approach like using openvpn http://www.voip-info.org/wiki/view/IAX_OpenVPN ? This half-encrypted iax encryption doesn't make much sense to me, therefore I think there's probably something I'm missing/misunderstanding. Best Regards, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ -- 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] wireless VOIP phone recommendations?
On Tue, 2008-02-05 at 14:37 +, [EMAIL PROTECTED] wrote: Hi, I use Linksys WIP 330 and the sound is good, talk time with full battery go up to 2 hours, I'm happy with. Ahhh. OP wanted to know about wirelessly networked phones. Interesting as they are (and expensive -- the WIP-330 retails for $229 at voiplink.com), I was hoping this would be a thread about simply cordless IP (SIP or IAX) phones. I think these tend to be available at a more reasonable price. I have a Panasonic GLOBALRANGE BB-GT1500CB (http://www.panasonic.ca/english/telecom/telephones/globarange/index.asp) which technically is supposed to only work with the Joip service, but spoofing this phone to work with your Asterisk server is not too difficult. It's a reasonable phone at a reasonable price (CAN$70 the last time I looked at a retail shop) but I have found that it can drop out sometimes. I tend to think the drop-out is in the audio handling in the handset itself rather than anything on the network. It seems like it might be some kind of silence detection and optimization circuitry (i.e. not transmitting dead air to the base station) that just doesn't work in real-life as well as it did on paper. Also this Panasonic phone does not do call-waiting. When there is a call in session on it, an attempt to route a second call to it from Asterisk results in a busy here message back from the phone. :-( I wonder what else is out there in a more affordable consumer price range. I guess there is always ATAs and regular phones. I've always wondered though if there is any benefit to even a basic phone such as the GLOBALRANGE phones being native SIP vs. just using an ATA. I have not discovered anything this phone can do above and beyond what our standard cordless Panasonic phone does plugged into an ATA. b. signature.asc Description: This is a digitally signed message part ___ -- 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] wireless VOIP phone recommendations?
Hi, I use Linksys WIP 330 and the sound is good, talk time with full battery go up to 2 hours, I'm happy with. Best regards, Chris Hariga Sent from my BlackBerry® wireless device -Original Message- From: marvin horst [EMAIL PROTECTED] Date: Tue, 5 Feb 2008 08:52:44 To:asterisk-users@lists.digium.com Subject: [asterisk-users] wireless VOIP phone recommendations? I have been using the D-Link DPH-540 wireless VOIP handset, and I really like this phone. We had tried the UStarcomm phone, but the phone is used in a noisy environment and the volume wasn't loud enough. The problem with the D-Link phone is the Li-ion battery needs to be replaced and D-Link doesn't sell a replacement battery and I haven't found any after-market batteries. So this phone is essentially a brick because I need a new battery :( So any recommendations for another wireless VOIP phone? -- Marvin Horst ___ -- 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] Zaptel timer on Intel Dual Core servers
Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Feb 05, 2008 at 09:58:54AM +, Tony Mountifield wrote: Specifically, what kernel version? As I understand from an IRC chat - CentOS 5 - 2.6.18-something. Hmm, ok. If you are using 2.6.9 (e.g. Centos 4), then by default when you build ztdummy, it does NOT use USE_RTC (although IMHO it should do). You should make the following change to ztdummy.c: #if LINUX_VERSION_CODE = VERSION_CODE(2,6,13) On a kernel version that calls itself 2.6.9, that part is never reached. So that point is mute. moot, not mute. But in fact, it isn't, since the #if's then part only goes as far as... #define USE_RTC ... here, ... #else ... and so we are here for kernel versions 2.6.13 -#if 0 +#if 1 #define USE_RTC #endif #endif Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- 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] wireless VOIP phone recommendations?
Brian J. Murrell wrote: On Tue, 2008-02-05 at 14:37 +, [EMAIL PROTECTED] wrote: Hi, I use Linksys WIP 330 and the sound is good, talk time with full battery go up to 2 hours, I'm happy with. Ahhh. OP wanted to know about wirelessly networked phones. Interesting as they are (and expensive -- the WIP-330 retails for $229 at voiplink.com), I was hoping this would be a thread about simply cordless IP (SIP or IAX) phones. I think these tend to be available at a more reasonable price. I have a Panasonic GLOBALRANGE BB-GT1500CB (http://www.panasonic.ca/english/telecom/telephones/globarange/index.asp) which technically is supposed to only work with the Joip service, but spoofing this phone to work with your Asterisk server is not too difficult. It's a reasonable phone at a reasonable price (CAN$70 the last time I looked at a retail shop) but I have found that it can drop out sometimes. I tend to think the drop-out is in the audio handling in the handset itself rather than anything on the network. It seems like it might be some kind of silence detection and optimization circuitry (i.e. not transmitting dead air to the base station) that just doesn't work in real-life as well as it did on paper. Also this Panasonic phone does not do call-waiting. When there is a call in session on it, an attempt to route a second call to it from Asterisk results in a busy here message back from the phone. :-( I wonder what else is out there in a more affordable consumer price range. I guess there is always ATAs and regular phones. I've always wondered though if there is any benefit to even a basic phone such as the GLOBALRANGE phones being native SIP vs. just using an ATA. I have not discovered anything this phone can do above and beyond what our standard cordless Panasonic phone does plugged into an ATA. b. ___ -- 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 For the most part, for day to day dialing, you won't see any really significant difference between a WiFi phone and an ATA with a regular cordless or DECT phone. You may lose the ability to dial SIP URIs (although not all wifi sip handsets have this ability). However, in general, none of the true Wi-Fi phones we've tested other than the Nokia E series have been worth mucking with. Dropping off APs, poor NAT capability, low battery life, troublesome configurations, random weirdness -- these seem to abound in the world of wi-fi SIP. This is why the usual scenario for any sort of office-wide deployment involves DECT. It's a shame, really. With wi-fi being so prevalent so many places we go, and with the possibility for portability being outstanding, it's a shame the hardware manufacturers haven't quite made anything worth buying. N. ___ -- 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] Zaptel timer on Intel Dual Core servers
On Tue, 5 Feb 2008, Tony Mountifield wrote: Tzafrir Cohen [EMAIL PROTECTED] wrote: So that point is mute. moot, not mute. moot, not mute. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] wireless VOIP phone recommendations?
However, in general, none of the true Wi-Fi phones we've tested other than the Nokia E series have been worth mucking with. Dropping off APs, poor NAT capability, low battery life, troublesome configurations, random weirdness -- these seem to abound in the world of wi-fi SIP. This is why the usual scenario for any sort of office-wide deployment involves DECT. My early experiments with wifi handhelds were not good. I might reevaluate based on some of the new dual mode phones from Nokia and RIM. Earlier this month I installed a set of the new Snom M3 SIP/DECT phones. I have two handsets and one base. While not cheap these are far, far better than any wifi phone I've ever used. I'm told that the Siemens VOIP capable models are cheaper and also very good . A full length review of the M3 will appear on www.smallnetbuilder.com in a few weeks. Michael Graves -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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] Higher level API on top of AMI and AGI (was Re: Real API for Perl?)
Stefan Reuter wrote: Lee Jenkins wrote: I thought that the OP was asking for something to perl what Asterisk-Java does for java coders. I would definitely consider Asterisk-Java to be a framework, though not so much with PasAGI which is more of an class object wrapper around AGI functions that I wrote a while back because I'm lazy that way ;) Indeed and I think such a higher level API could be implemented in different languages. There is/was a port of the Asterisk-Java API to .Net at least. I think especially the live API of Asterisk-Java is worth having a look at. It provides an object view on top of AMI with rich objects like Channel and methods like hangup() and redirect(). So it makes the developer focus on his tasks rather than thinking in terms of actions and responses. Asterisk 1.6 includes a new feature that allows using AMI as a transport for AGI commands, there abstraction becomes even more important. For Asterisk-Java I am currently adding support for that in a way that allows the developer to run the same AGI code either through FastAGI or AMI without knowing about the underlying details. Where is more information on this new feature for Asterisk 1.6? Any details? If someone is interested in defining a language-neutral general higher level API that can be implemented in a variety of languages I am happy to support this effort. This would be refreshing as the current AMI output is a little all over the place. Example: Conf Num PartiesMarked Activity Creation 1110001 0001 00:17:57 Dynamic Above is a line from MeetMe command issued from AMI. After the header line, each successive line denote information about a conference. No problems there, except there is an extraneous Tab (#9) character right after the Parties field which screws you up when parsing until you figure out that there is a Tab character there. There appears to be no reason to have a tab character there that I can see, well maybe to trip up unwary developers ;) I'm not sure what your point is, but I'll say that I'm a definite proponent of abstraction layers provided they don't bar access to lower level logic when I need it. I think you'll agree that good abstractions lend themselves to reuse and reduced development time (easy of use, less runtime logic errors, easier to extend, etc). And don't miss the additional benefit of supporting multiple versions of Asterisk that you get almost for free. Asterisk-Java will run with Asterisk from 1.0 to 1.6 without changing your code even if the Asterisk guys decide to rename properties and the like. Just have a look at doc/manager_1_1.txt in the betas of Asterisk 1.6 and decide what your efforts would be to support Asterisk 1.4 and 1.6 if you stick to low level APIs. Another great reason for abstraction/encapsulation IMO. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- 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] Telephony Migration Hardware
We are maxed out on our legacy PBX, and the question is the process of migrating to a new * system from the legacy. We current have 36 FXO lines coming into our site, and the usage on these lines indicates we can spare a few of them to launch the * server, and then move additional lines over as the * server/network/cable plant is built out and traffic moves to *. Eventually, we will want to merge all oustide traffic to SIP WAN circuit terminated on *. The question is what * telephony interface hardware to use for the migration. At the beginning we need a couple of FXO ports, and at the end we will want to have some kind digital trunking. Any recommendation for hardware to use throughout this migration process? Thanks a bunch! . ___ -- 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] Higher level API on top of AMI and AGI (was Re: Real API for Perl?)
Asterisk 1.6 includes a new feature that allows using AMI as a transport for AGI commands, there abstraction becomes even more important. For Asterisk-Java I am currently adding support for that in a way that allows the developer to run the same AGI code either through FastAGI or AMI without knowing about the underlying details. Where is more information on this new feature for Asterisk 1.6? Any details? I wrote this blog entry when I was writing the AsyncAGI feature: http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ This is the bug entry: http://bugs.digium.com/view.php?id=11282 I changed my mind regarding the behavior of this feature after opening the bug entry, so the initial description of the bug can be confusing and totally different from the final implementation and behavior, so you will have to read all the comments in the bug entry to understand what is this about. Moisés Silva -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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] meetme with ztxen - WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device
Hi, I have asterisk installed in the xen virtual server. I installed zaptel 1.4.2.1 and patched it to have ztxen module. I loaded ztxen module but when I try to invoke or call to my meetme application I get the following warning and negative result of connecting to conference: [Feb 5 17:46:13] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:13] -- SIP/sip.rd.touk.pl-b0006fc0 Playing 'conf-invalid' (language 'en') [Feb 5 17:46:17] -- SIP/sip.rd.touk.pl-b0006fc0 Playing 'conf-getconfno' (language 'en') [Feb 5 17:46:26] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:26] -- SIP/sip.rd.touk.pl-b0006fc0 Playing 'conf-invalid' (language 'en') [Feb 5 17:46:29] -- SIP/sip.rd.touk.pl-b0006fc0 Playing 'conf-getconfno' (language 'en') Any ideas what is wrong? Kind regards Tomasz ___ -- 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] Zaptel timer on Intel Dual Core servers
In article [EMAIL PROTECTED], Steve Edwards [EMAIL PROTECTED] wrote: On Tue, 5 Feb 2008, Tony Mountifield wrote: Tzafrir Cohen [EMAIL PROTECTED] wrote: So that point is mute. moot, not mute. moot, not mute. Now that point really *is* moot (i.e. open to debate, which is the true meaning of moot). I only put punctuation inside the quotes if the punctuation is part of what was being quoted, rather than part of the quoting sentence. See http://alt-usage-english.org/excerpts/fxvs.html Anyway, this is way OT, so I won't comment further. :-) Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- 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] Post Call QoS?
Ok, so I've asked this question before, and didn't get an answer. So here I go again! Asterisk 1.4 has some channel variables that you can inspect after a call is complete that will give you QoS metrics. Stuff like average round trip time, etc. Since there's only one set of variables, and calls will have two channels, which channel is this information for? Is it for one of the channels? Is it an aggregate of both channels? Who added this code and what where they thinking when they wrote it? Thanks, Doug. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs___ -- 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] What causes this?
I have found several references to this problem, but never a solution. I have fixed it before, but it was always by accident... Feb 5 13:27:39 NOTICE[12924]: channel.c:1904 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to slin Feb 5 13:27:39 NOTICE[12924]: channel.c:1904 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to slin Feb 5 13:27:39 NOTICE[12924]: channel.c:1904 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to slin Feb 5 13:27:39 NOTICE[12924]: channel.c:1904 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to slin Feb 5 13:27:39 NOTICE[12924]: channel.c:1904 ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to slin Obviously, I have obfuscated the real number with #... Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Attachment encrypted? click here http://www.highpoweredhelp.com/tutorials/wincrypt/ . ___ -- 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] Console/dsp, makes me sound like a Dalek
Thomas Kenyon wrote: The server that I will need to get this running on has an 82801EB/ER (ICH5/ICH5R) AC'97 sound controller (and no expansion space left to put another card in). Just a suggestion, don't forget there are USB audio devices available that work with linux, you may have an extra usb port ;) Not suggesting it because I've tried this for asterisk, just thinking outside the box :) ___ -- 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] GROUP_COUNT and Attended transfer
Hi Paul, Am Dienstag, den 05.02.2008, 10:10 +1100 schrieb Paul Hales: With some of the phones (snom, for example) you can turn off mwi, so the phone will only accept one call at a time. Much easier. PaulH Thanks for Your answer. Unfortunaly turning call waiting off is not an option for me. Some clients aren't able to switch it off and some users want to use the web gui to set the group count via the * database. Do You know, if it is a bug or a feature? Regards Karsten ___ -- 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] Telephony Migration Hardware
We are maxed out on our legacy PBX, and the question is the process of migrating to a new * system from the legacy. We current have 36 FXO lines coming into our site, and the usage on these lines indicates we can spare a few of them to launch the * server, and then move additional lines over as the * server/network/cable plant is built out and traffic moves to *. Eventually, we will want to merge all oustide traffic to SIP WAN circuit terminated on *. The question is what * telephony interface hardware to use for the migration. At the beginning we need a couple of FXO ports, and at the end we will want to have some kind digital trunking. Bit difficult to offer you useful advice without knowing which country you're in (and hence what the local telco will do for you free of charge), but I'll try and answer as if you were based in the UK. If your eventual target is to have all calls coming in via IP, I'd recommend one of the low-end Digium FXO cards (TDM400 with a couple of FXO modules). This will give you a couple of analogue channels for things like emergency services access etc. and avoid the need for you to register and (potentially) pay for PATS (aka E911 in the US). You'll then want to sort out whoever you're using for IP call termination and create them as a peer within asterisk. I'll assume you already know how to do that. Hopefully, whichever company you're using for inbound calls will provide you with a temporary number at this stage, which you can use to test call quality on inbound calls. Once you're satisfied the new server is behaving as it should, you can contact your analogue line provider and ask them to forward calls from your existing lines over to your temporary number from your IP provider. Certainly in the UK, although you'll be charged divert fees for calls to the number, there's no monthly charge for doing this. Give it a week or so like that to make sure everything's fine, during which time if any problems come to light, you can simply phone the telco and ask them to cancel the divert. After that, you should be able to port your number(s) on your analogue lines over to your IP trunk provider. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- 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] External MWI question for Asterisk
Gah. So currently in 1.4, there is no method of having Asterisk accept SIP NOTIFY from another server, and pass it on to endpoints if it matches? I can't imagine this being that complex, but then again I'm not familiar with the Asterisk internals. It just seems Asterisk would compare the SIP NOTIFY to what it has currently registered (sip show peers) and forward it on to the endpoint. I'm pretty sure sipXecs can do this. Anyway, thanks for the reply Olle. I think if I re-design my solution for the phones to register with sipXecs and not Asterisk I might make some headway, so that's my next move. On Feb 5, 2008 1:52 AM, Johansson Olle E [EMAIL PROTECTED] wrote: It is currently not possible. With the new event-driven MWI notification system in 1.6, it should be possible to add code for it, but it would be kind of tricky. If you send an MWI to an extension - how do we know where to send it? We either need to use the existing hints that connect the extensions to the device name space, or add a new sort of voicemail hints that connects an extension to a voicemailbox ID that we devices can subscribe to. /O ___ -- 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] wireless VOIP phone recommendations?
Michael Graves wrote: However, in general, none of the true Wi-Fi phones we've tested other than the Nokia E series have been worth mucking with. Dropping off APs, poor NAT capability, low battery life, troublesome configurations, random weirdness -- these seem to abound in the world of wi-fi SIP. This is why the usual scenario for any sort of office-wide deployment involves DECT. My early experiments with wifi handhelds were not good. I might reevaluate based on some of the new dual mode phones from Nokia and RIM. Earlier this month I installed a set of the new Snom M3 SIP/DECT phones. I have two handsets and one base. While not cheap these are far, far better than any wifi phone I've ever used. I'm told that the Siemens VOIP capable models are cheaper and also very good . A full length review of the M3 will appear on www.smallnetbuilder.com in a few weeks. For the record the best sound quality of any wifi/dect phone i've experienced so far is the humble BT Home Hub with hubphone 1010. Ok its not exactly wifi, but if you think of it as a dect SIP ATA with an optional AP (WDS compliant), adsl modem (its optional you can turn it off) and NAS usb client (it runs samba, again you can turn it off) plus a firewall/router (IPTABLES)its a great piece of kit. Also inexpensive, I pay around 10GBP inc p+p for these from ebay. The only drawback is you have to hack the xml config file as the webui is totally crippled by BT. All in all a very good phone, the hands free mode is the loudest/clearest/echo free phone i've ever had. Bails ___ -- 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 POlycom question
randulo wrote: On Feb 4, 2008 9:34 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In my recollection, [EMAIL PROTECTED] worked when I tried it, without sip or a colon. xxx could be anything at all. I noted this behavior back in 2006: http://lists.digium.com/pipermail/asterisk-users/2006-March/146393.html Note, that was with asterisk 1.2 I am running asterisk 1.2 although it shouldn't matter because I do not want to go thru asterisk (hence the OT) the number I put in the directory or dial in manually is of the style [EMAIL PROTECTED] (no colon or sip) ___ -- 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 For me, that worked fine back in 2006 exactly as you have it. I have url-dialing turned off right now so can't double-check. Sorry it's not working for you. There are quite a few places that could break IMO. On second thought, I tried another angle: I pointed the phone's microbrowser at a page containing the following: a href=tel://[EMAIL PROTECTED]Joe Smith/abr a href=tel://[EMAIL PROTECTED]John Smith/a And it worked like a charm. Moj ___ -- 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] How to hookup to cell phone for outbound calls?
Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply. Can any of the 3G usb devices out there offer outbound analogue calls (ie other than via voip)? Cheers Ed W ___ -- 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] Cannot hear voice through SIP Phone from one side
I have a asterisk server. Two SIP Soft XLites are connected to the server. I am able to make calls from one SIP Phones to the other SIP Phones and landlines successfully. The SIP Soft Phone on th eother side can hear my voice but I cannot hear their voice. They can call my local cell phone as well. Samething, they can hears my voice, I cannot hear their voice. The microphone and speakers are working on both sides because we are able to use google talk and are able to talk successfully. But it would not work on XLite over asterisk for some reason. The Asterisk server is a linux server. There is no firewall between the servers. It is in a DMZ. Any suggestion how to get it to work :) Thanks, Sanjoy. Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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] How to hookup to cell phone for outbound calls?
Ed W wrote: Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply. Can any of the 3G usb devices out there offer outbound analogue calls (ie other than via voip)? Cheers Ed W ___ -- 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 How about http://www.mgamble.ca/oss/iphone_asterisk/ ? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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] Cannot hear voice through SIP Phone from one side
On Feb 5, 2008 2:32 PM, Sanjoy Rath [EMAIL PROTECTED] wrote: The Asterisk server is a linux server. There is no firewall between the servers. It is in a DMZ. My bet is that it's not a *true* DMZ. You're still dealing with NAT, and that's what's causing the one-way audio. This topic has been discussed ad nauseam on the list and is documented quite well on the wiki - search there and you'll most likely find the answers you're looking for. -erik ___ -- 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] How to hookup to cell phone for outbound calls?
On Feb 5, 2008 2:37 PM, Drew Gibson [EMAIL PROTECTED] wrote: How about http://www.mgamble.ca/oss/iphone_asterisk/ ? Hah! Cool, but quite ridiculous. :-) ___ -- 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] Cannot hear voice through SIP Phone from one side
I have an asterisk server. Two SIP Soft XLites are connected to the server. I am able to make calls from one SIP Phones to the other SIP Phones and landlines successfully. The SIP Soft Phone on th eother side can hear my voice but I cannot hear their voice. They can call my local cell phone as well. Samething, they can hears my voice, I cannot hear their voice.The microphone and speakers are working on both sides because we are able to use google talk and are able to talk successfully. But it would not work on XLite over asterisk for some reason.The Asterisk server is a linux server. There is no firewall between the servers. It is in a DMZ.Any suggestion how to get it to work :)Thanks,Sanjoy. _ ___ -- 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] Mistake in the wiki's description of cmd Pickup() ?
Hi Stefan, Am Dienstag, den 05.02.2008, 10:30 +0100 schrieb Stefan Guenther: Hi, according to the description of Pickup() on page http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup I can use this command to pickup a call at a certain extensions. When I try this with e.g. exten = *8200,1,Pickup(200) I see, that You are using bristuffed *. As bristuff has its own pickup mechanism, be careful using the right one. AFAIK You have to use DPickup if You want to pickup a call by extension. In the bristuffed version of * Pickup is used with a group and DPickup is used with an extension (AFAIK). HTH, Karsten ___ -- 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] How to hookup to cell phone for outbound calls?
Erik Anderson wrote: On Feb 5, 2008 2:37 PM, Drew Gibson [EMAIL PROTECTED] wrote: How about http://www.mgamble.ca/oss/iphone_asterisk/ ? Hah! Cool, but quite ridiculous. :-) I have a Linksys NSLU2 (Slug) at home running Asterisk (see http://www.nslu2-linux.org/ ) It's small, relatively cheap and runs Asterisk very well. You could slip it into a pocket. I haven't tried yet but I have done a little reading and hope to connect the Slug to a mobile network when I get the time to play. I know you said no bluetooth, Ed but if you're in North America and your cellular network is CDMA, AFAIK option 1 is the only one possible. These carriers generally won't allow devices on their networks unless they are purchased from the carrier. If your cellular network is GSM then there are two approaches to try, 1. Slug, 4GB USB stick, USB Bluetooth dongle, dedicated bluetooth mobile phone + Asterisk 1.4 with a chan_mobile. Unfortunately, I doubt that chan_mobile is packaged for the slug (it's in 1.4 trunk) and you would have to build it. Cost ~$150 + phone + your time 2. Slug, 4GB USB stick + SIP-GSM gateway. Much easier to configure but it's a second box so less portable and more expensive than a BT dongle and an old phone. Probably more robust. Cost $250-$400 You could also substitute a Linksys WRT54GL ( http://openwrt.org/ ) for the Slug which would give you ethernet ports and wireless too. Hope this gives you some ideas regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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] Cannot hear voice through SIP Phone from one side
Erik Anderson wrote: On Feb 5, 2008 2:32 PM, Sanjoy Rath [EMAIL PROTECTED] wrote: The Asterisk server is a linux server. There is no firewall between the servers. It is in a DMZ. My bet is that it's not a *true* DMZ. You're still dealing with NAT, and that's what's causing the one-way audio. This topic has been discussed ad nauseam on the list and is documented quite well on the wiki - search there and you'll most likely find the answers you're looking for. Erik, i having the exact same problem, but couldn't find anything on the wiki for that. Maybe you could assist and point us newbies to the relevant page. Is there a mailing list archive search engine somewhere? 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] wireless VOIP phone recommendations?
So any recommendations for another wireless VOIP phone? As someone else pointed out, the Siemens C450 IP (and higher models) work great! Also, the snom m3 gets some good reviews and will be the next one I'll try out.. cheers, stoffell ___ -- 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] wireless VOIP phone recommendations?
A linksys PAP2 with a Motorola Dect set is what I use for a wireless IP phone solution. I have tried Zyxel y Linksys wifi phones, and a couple of others, but the battery life just isn't workable on WIFI phones. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de marvin horst Enviado el: martes, 05 de febrero de 2008 14:53 Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] wireless VOIP phone recommendations? I have been using the D-Link DPH-540 wireless VOIP handset, and I really like this phone. We had tried the UStarcomm phone, but the phone is used in a noisy environment and the volume wasn't loud enough. The problem with the D-Link phone is the Li-ion battery needs to be replaced and D-Link doesn't sell a replacement battery and I haven't found any after-market batteries. So this phone is essentially a brick because I need a new battery :( So any recommendations for another wireless VOIP phone? -- Marvin Horst ___ -- 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] wireless VOIP phone recommendations?
As someone else pointed out, the Siemens C450 IP (and higher models) work great! I should point out that for the relatively small price difference, it's well worth getting the S450 rather than the C460. The screen on the 's' series is much more crisp and higher resolution. If you use the thing regularly, you'll be grateful for the improvement. I'd like to get my hands on a Snom M3 to test, but over here in the UK it's nearly 3x the price of the Siemens S450,so I fear customer uptake will be limited at best. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ -- 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] External MWI question for Asterisk
- Original Message From: Jason Crum [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 5 February, 2008 7:13:12 PM Subject: Re: [asterisk-users] External MWI question for Asterisk Gah. So currently in 1.4, there is no method of having Asterisk accept SIP NOTIFY from another server, and pass it on to endpoints if it matches? I can't imagine this being that complex, but then again I'm not familiar with the Asterisk internals. It just seems Asterisk would compare the SIP NOTIFY to what it has currently registered (sip show peers) and forward it on to the endpoint. I'm pretty sure sipXecs can do this. Hi Jason, We use Asterisk with realtime so that all the SIP peer's contact URI's are recorded in a database. Our separate MWI service then is able to lookup where to send the MWI notifications to and doesn't need to involve Asterisk in their sending. Regards, Greyman. Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail ___ -- 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] wireless VOIP phone recommendations?
On Tue, 5 Feb 2008 22:35:38 -, Chris Bagnall wrote: As someone else pointed out, the Siemens C450 IP (and higher models) work great! I should point out that for the relatively small price difference, it's well worth getting the S450 rather than the C460. The screen on the 's' series is much more crisp and higher resolution. If you use the thing regularly, you'll be grateful for the improvement. I'd like to get my hands on a Snom M3 to test, but over here in the UK it's nearly 3x the price of the Siemens S450,so I fear customer uptake will be limited at best. Is the new Gigaset S675 IP actually available? And has anyone tried it? I can't find it available in the US. I'm wondering if it's worth waiting or should I just get one of the older models? Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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] External MWI question for Asterisk
Ah, so you have a MWI service that polls the Asterisk realtime DB for the SIP URI information for an external voicemail system? I'm guessing whatever you're using for voicemail alerts the your MWI service (custom written for this?)? On Feb 5, 2008 5:36 PM, Grey Man [EMAIL PROTECTED] wrote: Hi Jason, We use Asterisk with realtime so that all the SIP peer's contact URI's are recorded in a database. Our separate MWI service then is able to lookup where to send the MWI notifications to and doesn't need to involve Asterisk in their sending. Regards, Greyman. Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail ___ -- 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] Enterprise or Fedora?
Oh, yes, asterisk is in universe, but i prefer apt-get build-dep asterisk And then compile from current source, not the package in universe -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Tzafrir Cohen Enviado el: Sábado, 02 de Febrero de 2008 06:17 p.m. Para: asterisk-users@lists.digium.com Asunto: Re: [asterisk-users] Enterprise or Fedora? On Sat, Feb 02, 2008 at 05:13:39PM -0600, [EMAIL PROTECTED] wrote: Ubuntu server for me please simply, is better... install. then activate universe and multiverse repositories sudo apt-get update sudo apt-get upgrade sudo apt-get build-dep asterisk and then... tar xvfz ./configure make make install ... is very very easy and clean, and IMHO i guess is better SO Ubuntu than any other RHEL based distro... Mind you, asterisk is in the ubuntu universe (rather than main) archive. As such, it also has some other dependencies in universe. Specifically (looking at the current Hardy) package: libc-client2007-dev libiksemel-dev libopenh323-dev libpri-dev libradiusclient-ng-dev libtonezone-dev libvpb-dev zaptel-source As a rule, the universe packages there get much less attention, and don't necessarily get regular security updates. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 __ Information from ESET Smart Security, version of virus signature database 2849 (20080205) __ The message was checked by ESET Smart Security. http://www.eset.com __ Information from ESET Smart Security, version of virus signature database 2851 (20080205) __ The message was checked by ESET Smart Security. http://www.eset.com ___ -- 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] Can't delete voicemail messages
Hi list, After recently setting up voicemail for Asterisk 1.4.14 on my Debian etch server, I noticed that I can't delete any old voicemail messages. The voicemail menu option Press 7 to delete this message is available, but when I press 7 the response is always message undeleted and the message is still there. What could I be missing here? Thanks, Jaap ___ -- 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] Can't delete voicemail messages
On 00:38, Wed 06 Feb 08, Jaap Winius wrote: Hi list, After recently setting up voicemail for Asterisk 1.4.14 on my Debian etch server, I noticed that I can't delete any old voicemail messages. The voicemail menu option Press 7 to delete this message is available, but when I press 7 the response is always message undeleted and the message is still there. What could I be missing here? Can you post the CLI logs from when that is happening ? -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] Console/dsp, makes me sound like a Dalek
Mojo with Horan Company, LLC wrote: Thomas Kenyon wrote: The server that I will need to get this running on has an 82801EB/ER (ICH5/ICH5R) AC'97 sound controller (and no expansion space left to put another card in). Just a suggestion, don't forget there are USB audio devices available that work with linux, you may have an extra usb port ;) Not suggesting it because I've tried this for asterisk, just thinking outside the box :) Thanks, I never did get it working at home, but on the server I need it for it worked first time. (same method). Thanks for the replies. ___ -- 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] Can't delete voicemail messages
Quoting Michiel van Baak [EMAIL PROTECTED]: On 00:38, Wed 06 Feb 08, Jaap Winius wrote: Hi list, After recently setting up voicemail for Asterisk 1.4.14 on my Debian etch server, I noticed that I can't delete any old voicemail messages. The voicemail menu option Press 7 to delete this message is available, but when I press 7 the response is always message undeleted and the message is still there. What could I be missing here? Can you post the CLI logs from when that is happening ? All I see is a list of sound files appearing as they are played -- no error messages of any kind. However, the sound files that are listed immediately after I hit 7 are: -- SIP/1000-081fc028 Playing 'vm-deleted' (language 'en') -- SIP/1000-081fc028 Playing 'vm-undeleted' (language 'en') I only hear the second one. These are quickly followed by a list of the usual menu options (see the full CLI log below involving this same call). Cheers, Jaap =Begin CLI log== == Spawn extension (phones-j, 7000, 6) exited non-zero on 'SIP/1000-081fc028' -- Executing [EMAIL PROTECTED]:1] Answer(SIP/1000-081fc028, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/1000-081fc028, 1) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/1000-081fc028, [EMAIL PROTECTED]|s) in new stack -- SIP/1000-081fc028 Playing 'vm-youhave' (language 'en') -- SIP/1000-081fc028 Playing 'digits/5' (language 'en') -- SIP/1000-081fc028 Playing 'vm-Old' (language 'en') -- SIP/1000-081fc028 Playing 'vm-messages' (language 'en') -- SIP/1000-081fc028 Playing 'vm-onefor' (language 'en') -- SIP/1000-081fc028 Playing 'vm-Old' (language 'en') -- SIP/1000-081fc028 Playing 'vm-messages' (language 'en') -- SIP/1000-081fc028 Playing 'vm-opts' (language 'en') -- SIP/1000-081fc028 Playing 'vm-helpexit' (language 'en') -- SIP/1000-081fc028 Playing 'vm-first' (language 'en') == Parsing '/var/spool/asterisk/voicemail/default/1000/Old/msg.txt': Found -- SIP/1000-081fc028 Playing '/var/spool/asterisk/voicemail/default/1000/Old/msg' (language 'en') -- SIP/1000-081fc028 Playing 'vm-advopts' (language 'en') -- SIP/1000-081fc028 Playing 'vm-repeat' (language 'en') -- SIP/1000-081fc028 Playing 'vm-next' (language 'en') -- SIP/1000-081fc028 Playing 'vm-delete' (language 'en') -- SIP/1000-081fc028 Playing 'vm-toforward' (language 'en') -- SIP/1000-081fc028 Playing 'vm-savemessage' (language 'en') -- SIP/1000-081fc028 Playing 'vm-helpexit' (language 'en') -- SIP/1000-081fc028 Playing 'vm-deleted' (language 'en') -- SIP/1000-081fc028 Playing 'vm-undeleted' (language 'en') -- SIP/1000-081fc028 Playing 'vm-advopts' (language 'en') -- SIP/1000-081fc028 Playing 'vm-repeat' (language 'en') -- SIP/1000-081fc028 Playing 'vm-next' (language 'en') -- SIP/1000-081fc028 Playing 'vm-delete' (language 'en') -- SIP/1000-081fc028 Playing 'vm-toforward' (language 'en') -- SIP/1000-081fc028 Playing 'vm-savemessage' (language 'en') -- SIP/1000-081fc028 Playing 'vm-helpexit' (language 'en') -- SIP/1000-081fc028 Playing 'vm-goodbye' (language 'en') -- Executing [EMAIL PROTECTED]:4] Wait(SIP/1000-081fc028, 1) in new stack -- Executing [EMAIL PROTECTED]:5] Playback(SIP/1000-081fc028, vm-goodbye) in new stack -- SIP/1000-081fc028 Playing 'vm-goodbye' (language 'en') -- Executing [EMAIL PROTECTED]:6] Hangup(SIP/1000-081fc028, ) in new stack == Spawn extension (phones-j, 7000, 6) exited non-zero on 'SIP/1000-081fc028' =End CLI log ___ -- 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] External MWI question for Asterisk
- Original Message From: Jason Crum [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 5 February, 2008 10:59:59 PM Subject: Re: [asterisk-users] External MWI question for Asterisk Ah, so you have a MWI service that polls the Asterisk realtime DB for the SIP URI information for an external voicemail system? I'm guessing whatever you're using for voicemail alerts the your MWI service (custom written for this?)? Hi Jason, We use the Asterisk voicemail system as well and have a custom app that accepts the piped voicemail from Asterisk, saves it and then sets a flag in the realtime database that the MWI service can check. It sounds like a lot of bits and pieces but it all works well. Regards, Greyman. Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail ___ -- 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] How to hookup to cell phone for outbound calls?
Well I think you need a GSM Gateway You can find some info on cyber-telecom.net For a cheap option you can try a CT-G1000 or CT-G2000 and then plug it in a X100P or something similar then it would be very economical. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed W Sent: Wednesday, February 06, 2008 4:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to hookup to cell phone for outbound calls? Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply. Can any of the 3G usb devices out there offer outbound analogue calls (ie other than via voip)? Cheers Ed W ___ -- 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] Gemeinschaft released
Hi, Just wanted to let you know that we have just made our GPL toolkit Gemeinschaft available to the public. (Finally.) Mostly German for now - about half of the strings in the language strings file have been translated to English. I'm a software developer, not a marketing guy, so ... svn co https://svn.amooma.de/gemeinschaft/trunk gemeinschaft-trunk German readers: see http://www.amooma.de/gemeinschaft/ Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- 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] [Softphones] ZoIPer vs. XLite?
Marc, does your client play nicely with Vista? We've been having some problems with softphones that work fine in XP, but choke in Vista. I don't know, never tried it since I couldn't find a machine with enough power to run Vista decently ;) Try it and let me know how it goes. If it doesn't work, I will try to fix it. 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] Can't delete voicemail messages
Quoting Andy Doss [EMAIL PROTECTED]: File permission error? That is just my first guess. I am kind of new to Asterisk myself. The files are all in /var/spool/asterisk/voicemail/ where the asterisk user has read/write access to everything. Also, I see no error messages that would indicate a permission or access error. Thanks anyway, Jaap ___ -- 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] R2 with Alestra in Mexico...
I am trying to set up Astunicall 1.4.16 with a link from Alestra in Mexico City. I have done everything I usually do for other links in Mexico but this one simply will not send or receive calls. I just get Protocol error. Anyone has any experience with R2 and Alestra? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] Higher level API on top of AMI and AGI (was Re: Real API for Perl?)
Moises Silva wrote: Asterisk 1.6 includes a new feature that allows using AMI as a transport for AGI commands, there abstraction becomes even more important. For Asterisk-Java I am currently adding support for that in a way that allows the developer to run the same AGI code either through FastAGI or AMI without knowing about the underlying details. Where is more information on this new feature for Asterisk 1.6? Any details? I wrote this blog entry when I was writing the AsyncAGI feature: http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/ This is the bug entry: http://bugs.digium.com/view.php?id=11282 I changed my mind regarding the behavior of this feature after opening the bug entry, so the initial description of the bug can be confusing and totally different from the final implementation and behavior, so you will have to read all the comments in the bug entry to understand what is this about. Moisés Silva Thanks. That's pretty slick. It could add some flexibility, but as you noted on your blog, you could just as easily redirect to a FastAGI server, etc. Being able to call AGI's on a channel through the AMI seems like it could have some possibilities as well. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- 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] switch QOS requirements
Very Nice! Its much more reliable than translating DSCP to COS by switch which i'm not sure which switch does that and which one doesn't, and how they do it considering some gray area when you translate from DSCP to COS. On Feb 4, 2008 5:26 PM, Jared Smith [EMAIL PROTECTED] wrote: On Sun, 2008-02-03 at 22:42 -0700, Al lists wrote: Theoretically, setting TOS value ( these days called DSCP) wont change anything in switch behavior, unless you are using Layer 3 switches. What makes a difference in a switch is COS bits, and i'm not sure how asterisk sets that. In Asterisk 1.6, you will be able to set both the COS and TOS values. The sample sip.conf in the Asterisk 1.6 betas contains the following, to show you just how much you can adjust things :-) ;tos_sip=cs3; Sets TOS for SIP packets. ;tos_audio=ef ; Sets TOS for RTP audio packets. ;tos_video=af41 ; Sets TOS for RTP video packets. ;tos_text=af41 ; Sets TOS for RTP text packets. ;cos_sip=3 ; Sets 802.1p priority for SIP packets. ;cos_audio=5; Sets 802.1p priority for RTP audio packets. ;cos_video=4; Sets 802.1p priority for RTP video packets. ;cos_text=3 ; Sets 802.1p priority for RTP text packets. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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] GXV-3000 IP Video Phone
Hello All, I have 2 new Grandstream GXV 3000 phones and want to sell them to someone who is interested to buy. I can sell $200 per piece. If you are interested please reply this mail. Thanks, Thameem On May 8, 2007 7:25 AM, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello, So far yes... The Video phones are behaving good and all the functionality working. I have 5 phone on the network and planning to put more by next week. Cheers, Nitesh Noah Miller wrote: Hi Nitesh - Thanks everyone... The GXV-3000 IP Video Phone works with Asterisk 1.2 using H.263 Video Coder. I had to update both phones firmware with new one... Out of curiosity - do you like the phone? I've looked for reviews, but I haven't found any that rate the phone's functionality. - Noah ___ --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 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] External MWI question for Asterisk
Hi, 2008/2/5, Grey Man [EMAIL PROTECTED]: - Original Message From: Jason Crum [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, 5 February, 2008 7:13:12 PM Subject: Re: [asterisk-users] External MWI question for Asterisk Gah. So currently in 1.4, there is no method of having Asterisk accept SIP NOTIFY from another server, and pass it on to endpoints if it matches? I can't imagine this being that complex, but then again I'm not familiar with the Asterisk internals. It just seems Asterisk would compare the SIP NOTIFY to what it has currently registered (sip show peers) and forward it on to the endpoint. I'm pretty sure sipXecs can do this. Hi Jason, We use Asterisk with realtime so that all the SIP peer's contact URI's are recorded in a database. Our separate MWI service then is able to lookup where to send the MWI notifications Do you send those notifications to SIP hardphones ? Then, how do you proceed ? Is there a standard way to make (or stop) a SIP hardphone Message Waiting Indicator blinking ? Cheers to and doesn't need to involve Asterisk in their sending. Regards, Greyman. Get the name you always wanted with the new y7mail email address. www.yahoo7.com.au/y7mail ___ -- 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
