[Asterisk-Users] Asterisk LAMP Developer
_Description_ We are looking for an expert LAMP (Linux, Apache, MySQL, Perl, and PHP) developer with some Asterisk experience who is based in Western or Eastern Europe or Asia. We can work with an individual or an organization. You must be fluent in English. We need you to help expand development of Signate's core software products. As part of the Signate development team, you will design, develop, debug and test code for new and existing systems. You will work independently. You will be given a great deal of freedom to exercise your knowledge and experience. In return, we will expect high quality, elegant, well engineered solutions that meet the needs of our staff and customers and are delivered on time. Responsibilities include, but are not limited to: * Serve as a full-time LAMP developer. * Maintain and enhance our internal systems. * Work closely with other developers on the Tech-Dev team. * Perform time-zone based 7/24 support for our customer installations Requirements: * Ability and desire to learn independently as well as in a team * Ability to manage time well and meet multiple deadlines in a sometimes hectic environment. * Excellent writing and communication skills (English) To apply please send your CV and contact details to [EMAIL PROTECTED] _About Signate_ Signate is a leading global provider of design, installation, configuration, training and management services for open source VoIP telephony systems. For more information, visit Signate at http://www.signate.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco SIP Files
Asterisk and Cisco 79XX series configuration: http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx Christopher Jacob wrote: I am in the process of ordering a support contract from Cisco for my new 7960 phone, but I would really like to get it up and running. At the risk of being flamed off this list, could someone send me or point me in the direction of the SIP image files I need to change the phone over? Thanks, ~c ___ 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 -- dominique kull taridium.communications ltd t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 515151 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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] Voicemail forward to a remote server?
Anybody ever managed to implement a solution where one could forward a voicemail from one * server to another? Dominique ___ 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
Re: [Asterisk-Users] Voicemail forward to a remote server?
This certainly works, if you want to have a remote VM - but still does not forward a received VM to another server. Dominique Matthew Boehm wrote: Just have the two * servers login to eachother via IAX, then in your extensions plan where you normally have: exten = 8899,1,Dial(SIP/8899,15,tr) exten = 8899,2,Voicemail([EMAIL PROTECTED]) change it to exten = 8899,1,Dial(SIP/8899,15,tr) exten = 8899,2,Dial(IAX2/servername/extension) We have two * servers setup, #1 as main and #2 as pure MeetMe. We use the above IAX2 dial command to forward all calls on a paticular number to the other server. Works great. Matthew - Original Message - From: Dominique Kull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 21, 2004 9:56 AM Subject: [Asterisk-Users] Voicemail forward to a remote server? Anybody ever managed to implement a solution where one could forward a voicemail from one * server to another? Dominique ___ 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 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 -- dominique kull taridium.communications ltd t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 515151 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] tT funktions
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
You will find nearly all your questions answered in there.
Google search term:
site:voip-info.org asterisk dial command
and click I am feeling lucky :-)
Thomas Kuepper wrote:
(SIP/${EXTEN:[EMAIL PROTECTED],60,tT)
can anyone tell me what the tT attribute behind 60 stands for?
thx
--
Thomas Küpper
--
dominique kull
taridium.communications ltd
t: +44 207 731 1562
f: +44 207 900 6564
v: fwd 515151
w: http://taridium.com
e: [EMAIL PROTECTED]
___
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Re: [Asterisk-Users] Pingtel and some chinese company
Would be strange if it supported SIPRTP but not UDP... I think that most SIP support at least 10 protocols: SIP SDP RTP UDP DNS TFTP DHCP TCP IP ARP 802.3 (Ethernet) ;-) Dominique Mike Reed wrote: 1) Who bought Pingtel's phone line? 2) Anyone seen this chinese-made VoIP phone that supports 8 different protocols? http://www.telecom.globalsources.com/GeneralManager?language=endesign=cleanaction=GetArticlearticle_id=900055338page=printarticleprintThis=yes http://www.telecom.globalsources.com/GeneralManager?language=endesign=cleanaction=GetArticlearticle_id=900055338page=printarticleprintThis=yes Mike :) ___ 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
Re: [Asterisk-Users] Convert Cisco 7960 to sip
Don't worry - the downgrade is pretty painless. Just change the config to load the old firmware. Dominique Joel Vandal wrote: All Cisco 7940 that I have upgrade to 7.1 no more try to get the dialplan and ringlist files from tftp. Now I must found a way to downgrade from 7.1 to 6.3. -- Joel - Original Message - From: Simon Brown [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Convert Cisco 7960 to sip I've been using V7 for a couple of months now with no problems. Simon Brown ___ 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 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
Re: [Asterisk-Users] WiFi phone radiation regulation?
I have one of those SMC High Power W-LAN cards for special war driving applications :-) It is rated at 200mW and is sold over the counter. Ok, I usually don't put the card on my ear... Leo Ann Boon wrote: All, I just had the fortune to take one of the new Senao Wifi SIP phones for a short test drive. First look - it's a nice, compact phone. Weighs around 87g and roughly the size of a Nokia 6210. More on the those later. The thing that struck me was the RF power, it's rated at 100mw (20dBm). That's 10 times more than any of the other brands out on the market Cisco, WiSIP, Zyxel are all rated at 10mW. I'm not really sure if I want to stick something with that power to my ear. Assuming a reasonable antenna gain of 2.2dBi, we're talking about 22.2dBm - that's nearly 200mW of power radiating out of the phone. At 2.4GHz, it has higher penetration power than cells phones. My question: Does anyone know if cell phone SAR rules apply to WiFi phones as well? Over here in Singapore, there seems to be a loophole. As long as your equipment is in 2.4GHz, approval is not required if Tx power is 200mW. ___ 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 -- dominique kull taridium.communications ltd t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 515151 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Asterisk in a DMZ
Why not use a public address for * ? A firewall, if properly configured can protect your * server the same way as it would with NAT in a DMZ. Dominique Bastian Schern wrote: Hello *, I try to establish a Asterisk-Server for internal and external usage. Perfect use case for a DMZ, or not? My configuration: I N T E R N E T | | | E | | X | | T | | E | 213.xxx.xx.68 | R +-#+| N | Firewall || +-#+ - - - - - - - - - - - - - - - - - - - -+- | 192.168.40.68 | | | +#+| | Switch || +--#---#---#---#--+| | | | | +-+ | D | | | M +--+ | | Z | (213.xxx.xx.66) | (213.xxx.xx.70) | | 192.168.40.66| 192.168.40.70 | +-#+ +-#+| | Firewall | | Asterisk || +--+ +--+| | Server | | +-#+ - - - - - - - - - - - - - - - - - - - - -+- | 192.168.0.1| || +--+ | | | +#+| | Switch || I +--#--#--#--#--#--+| N | | | | T | | | | E | | | | R | | | | N | | +-+ | | +--+ | | | | | | | 192.168.0.101 | 192.168.0.102 | 192.168.0.103 | +--#---+ +--#---+ +--#---+ | | Tel1 | | Tel2 | | Tel3 | | +--+ +--+ +--+ | But now the IP-Phones could not communicate with Asterisk because the Server (a Linux host) will NAT the internal IP-Addresses. Is there a good way to solve this Problem? Regards Bastian ___ 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 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
Re: [Asterisk-Users] Wireless SIP Phones
Bodo Hahnke wrote: Hello, I found serveral discussions about the Zyxel ePhone Prestige P2000W and the WiSip from Pulver Innovations on this mailings list but still have some questions: 1) are there other affordable wireless SIP Phones on the market? I haven't seen or found anything else till now ... AFAIK not in large quantities - seen some other phones on paper, don't know if they exist though... 2) is p2000w and wisip the same hardware?? so could I use firmware from both companies regardless of what phone I buy?? The firmwares of the p2000W and the WiSIP are interchangeable. So yes. 3) does any of these phones have major bugs or will it be usable in a productive environment without getting mad or sleepless ?? The P200W and the WiSIP have quite a lot of bugs and usability issues. I would not use them in a productive environment unless it is with people with a technical background. The firmware and the phone are not there yet. The only Wireless SIP phone I would use in a productive environment would be the Cisco 7920. 4) any security issues with these phones? WEP is NOT secure - if you need security use wired and encrypted communication. Last but not least does anyone who knows both phones recommend anyone of these?? Or should I just buy the cheaper one? Buy the one you like better and use the WiSIP firmware on Monday's and Wednesday's then change it to ZyXEL for the rest of the week. :-) hope this helps Dominique ___ 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
Re: [Asterisk-Users] Wireless SIP Phones
You are right, there is no SIP firmware for the 7920 - SCCP is currently the only choice for *. Ray Burkholder wrote: yet. The only Wireless SIP phone I would use in a productive environment would be the Cisco 7920. I don't see a SIP load for the 7920. Are you sure it is SIP enabled? Ray. - This mail sent through IMP: http://horde.org/imp/ ___ 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
Re: [Asterisk-Users] WiSIP and Zyxel Prestige 2000W
How is the sound quality? Have you ever used a BT headset?? I have wasted too much money on BT in the last couple of years. If it still uses analog transmission for audio, I would skip this. I know there are digital audio devices around... but you still have the normal BT bandwidth limitations. But if you look at both Skype and Bluetooth you see that it is a very natural 'pairing' - Don't believe the Sk(h)ype ;-) If you need the serious stuff go with the Cisco 7920. cheers Dominique Dean Collins wrote: Bluetooth between the usb plug in the back of the pc and the handset. Yep the handset runs linux (or a derivative) I've been playing with the development hardware kit for the last week it runs your skype buddy list on the handset. Do a search on the asterisk email list, I wrote up a whole piece on this about 2 months ago saying that people should look into this as a way of enabling cordless handsets on asterisk. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Galt Sent: Thursday, 15 July 2004 2:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] WiSIP and Zyxel Prestige 2000W On Thu, 15 Jul 2004 10:39:23 +1000, Dean Collins [EMAIL PROTECTED] wrote: Lenz, can I suggest you check out the siemens gigaset skype cordless handset. It uses blue tooth bluetooth or 802.11? and linux on the handset to offer cordless capability. Do the HANDSET(s) run linux? could you provide a link for this and the sdk? I think that there should be an abaility to offer asterisk customisation on this product (there is a sdk on the website for free download). I'm not the person to do this odification but something worth looking at. Cheers, Dean ___ 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 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 -- dominique kull taridium.communications ltd t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] WiSIP and Zyxel Prestige 2000W
I can confirm that WEP with Netgear's ME103 is no problem. Latest firmware I found was here: http://www.zyxel.co.uk/support/ukadslfw.php joachim wrote: I'm also using a ZyXEL, and sound quality is very very bad when using WEP :/ Any solutions to this problem ? (or download links to newer/other firmware ?) -- dominique kull taridium.communications ltd t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] WiSIP and Zyxel Prestige 2000W
The ZyXEL firmware is still quite buggy and has some serious usability issues. When it works, it works quite well, sound quality is pretty ok with G.729. I tested the Pulver firmware, but did not notice any substantial difference (e.g. same bugs as with ZyXEL) I am still waiting for a 1.0 release. Go for it if you can live with early adopter pains. There is one bug which I really have a problem with: The phone does not properly communicate a SIP cancel to Asterisk. Some people claim that it works for them... Dominique Steve wrote: Hi, Anyone have any experience with either of these, I 'd appreciate some feedback? Plus it seems pretty easy to steal a connection with this. Zyxel Prestige 2000W WiSIP thanks, - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -- dominique kull taridium.communications ltd t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] HOW ASTERISK WORKS
http://www.voip-info.org/ see you in two months ;-) Giscard Fernandes Faria wrote: Hy guys, I cannot understand How the asterisk works. I would like know how the h323.conf, sip.conf and extension.conf works. I don't understand the parameters and the [sections]. What I need to the asterisk get a SIP call and forward them to a H323 terminal. I working at the h323.conf and extension.conf but I cannot understand!!! Please someone can help me. I your can send me a example (with comments) of a simple example working with sip and h323. Thanks. Giscard ___ Yahoo! Mail agora com 100MB, anti-spam e antivírus grátis! http://br.info.mail.yahoo.com/ ___ 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 -- dominique kull taridium.communications ltd t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Again on the ZyXEL Prestige 2000W
It is. I did a cross upgrade with Pulver's firmware. I could not notice any improvements, though... I still had that annoying hangup problem. lenz wrote: I have heard that the 2000W is the same exact harware as the PulverInnovations WiSip phone - http://www.pulverinnovations.com/ - so the drivers might be the same, but I have not tried this. dominique kull taridium.communications the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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] ZyXEL Prestige 2000W and DTMF
I've just seen this post: http://www.mail-archive.com/[EMAIL PROTECTED]/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is set to use DTMF 'outband' any ideas? Dominique -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] ZyXEL Prestige 2000W and DTMF
I have tested the exact same config, but had no luck. I managed to get it going with some different settings on the phone, though. ZyXEL settings: DTMF RELAY inband(RFC2833) ?? DTMF Payload 101 ?? for the sip.conf (same as Giles apart from forcing g.729) [400] type=friend username=400 secret=blah host=dynamic context=local dtmfmode=rfc2833 disallow=all allow=g729 callerid=Vintage Cell Phone 400 It is all a bit confusing regarding what is inband and outband on the phone. I am also not sure about DTMF Payload type... but it seems to work ok. regards Dominique Giles Scott wrote: Hi, With my config (as posted this morning) DTMF works. I can log onto voicemail by selecting a mailbox number and password Giles - Original Message - From: Dominique Kull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 24, 2004 12:02 PM Subject: [Asterisk-Users] ZyXEL Prestige 2000W and DTMF I've just seen this post: http://www.mail-archive.com/[EMAIL PROTECTED]/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is set to use DTMF 'outband' any ideas? Dominique -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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
Re: R: R: R: [Asterisk-Users] How to force G729
Did you try having two sip.conf entries for your gateway? Forcing one with G729 and the other with ulaw? You would obviously need to change your dialplan accordingly and have each phone configured so that it would take the proper extension. I have not tried this, it is just really an idea... Manuel Wenger wrote: If I understood your initial objective correctly (and I may not have), the user's phones are negotiating the codec to be used for each rtp session. Asterisk parameters can be used to dictate rtp sessions between the sip phone and asterisk, but that won't influence the next step in which the sip phone negotiates a new rtp session directly with the gateway. The gateway and the phone will negotiate a common codec based on whatever logic those two devices have been programmed with by their respective manufacturers; asterisk isn't involved. So, it sounds like the issue is understanding the codec selection logic that has been programmed into the gateway and the phone. I think you're getting my point, at least I think so (I'm getting more and more confused myself about this...) The problem is that the phone negotiates a codec with asterisk when placing the call (remember I have all reinvite's set to no, so the gateway and the phone won't talk directly to each other!). This negotiation actually works correctly, because I force the phone's codec using disallow=all; allow=g729 in the SIP phone's peer configuration. The negotiation which doesn't work the way I want is the asterisk-to-gateway part. The gateway can talk either ULAW or G729, whatever I tell it, if I force it using the disallow/allow method in sip.conf. The problem is that I need asterisk to talk to the gateway sometimes with ULAW, sometimes with G729, depending on the SIP phone who placed the call in the first place. What I need is some sort of command which says OK, now Dial(... @gateway), but force G729 (which works *if* I tell asterisk that the gateway supports G729 *only* in sip.conf, but we want it to support both codecs, right?). Apparently I can only force the codec on incoming channels, not on outgoing channels. Is this really an asterisk limitation? -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ 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 -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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
Re: R: [Asterisk-Users] How to force G729
Hmmm, I was thinking about this problem too... What type of gateway are you using? Is it registering with the Asterisk server? I would try using two different 'virtual' extensions on the gateway and in sip.conf. That way you would have full control on how calls from the gw to * are handled. Manuel Wenger wrote: That's actually a very good idea, and I have tried it: for outgoing calls it works like charm. But then the problem is transferred to incoming calls (from the gateway-asterisk-SIP client). Because the gateway now has 2 entries, asterisk is confused about what codec it has to use for incoming calls, and for some reason I can't force it, because the 2 entries have the same IP. I'm starting to think that I won't be able to solve that myself, but that someone will have to program something for this to work... But if I'm the only one having this kind of request, I'm not too optimistic -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ 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 -- taridium.communications dominique kull, partner the old lodge, london sw6 6ee uk t: +44 207 731 1562 f: +44 207 900 6564 v: fwd 268167 w: http://taridium.com e: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] C7960 g729 question
What does your sip.conf look like? Always make sure that you have the following codec order for G.729 pass-thru: [general] disallow=all allow=g729 allow=ulaw allow=alaw you don't need to force your C7960 (SIP settings) to use G.729 with the above config. see also: http://www.voip-info.org/tiki-index.php?page=Asterisk%20G.729%20pass-thru Dominique Rich Adamson wrote: I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 - g729 - asterisk - g711 - C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this? ___ 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 -- Dominique Kull The Old Lodge, London SW6 6EE UK t: +44 207 731 1562 v: fwd 268167 e: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] XML How To for Cisco 7960
Do a search on http://www.voip-info.org/ first. It is the best place for Asterisk and related stuff. http://www.voip-info.org/tiki-index.php?page=Asterisk%20Cisco%2079XX%20XML%20Services cheers Dominique Matthew John Darnell wrote: Aloha, Has anyone written an XML application for a Ciso 7960 phone running SIP? I can't find any examples anywhere! Anyone know of any resources for this? I have read it can render XML can get input from the keypad softkeys. Aloha, Matt ___ 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
Re: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails
been playing around with the Pulver firmware WF.00.11/B.00.13/Apr 07 2004 and its not better in any way. Anbody made some progress with that issue? I guess we will have to wait for ZyXEL releasing a real production FW. cheers Dominique Dominique Kull wrote: Thanks for your replies. The hangup is still failing with the latest CVS head. It seems to be a firmware issue. I am running WJ.00.0a / B.00.13 / Apr 12 2004 - Is there any newer release of the firmware floating around? cheers Dominique PS: Another interesting effect(IMHO bug): I cannot access the web interface after some time unless I make a call first. The same applies for pinging the handset. It only will reply after call has been established. Might be a power save feature... :-) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Pertti Pikkarainen Gesendet: Donnerstag, 3. Juni 2004 08:26 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails Prestige 2000W is the same BCM phone that was earlier referred as Wifi-600 in this list. http://www.bcm.com.tw/product/pdf/pdf1/Spec-WiFi600_2003_1103.pdf It has the same problem. If you enable WEP encryption ( 104 bit ), the voice becomes very choppy. Almost unusable. Without WEP it is fine. I wonder if anybody has better results with WEB enabled and with latest software releases ? -- Pertti Lars Boegild Thomsen wrote: I have noticed this one and I have also informed ZyXEL, but their response was vague to say the least. It is correct that the ZyXEL phone does not send a SIP Cancel when you disconnect an outgoing call that has not yet been picked up by the remote end. I have several times asked ZyXEL to put a formal bug report procedure in place with proper tracking but to no avail. Regards, Lars... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dominique Kull Sent: 02 June 2004 22:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying since it always leaves an empty VM. thanks Dominique ___ 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 -- Dominique Kull The Old Lodge, London SW6 6EE UK t: +44 207 731 1562 e: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails
Thanks for your replies. The hangup is still failing with the latest CVS head. It seems to be a firmware issue. I am running WJ.00.0a / B.00.13 / Apr 12 2004 - Is there any newer release of the firmware floating around? cheers Dominique PS: Another interesting effect(IMHO bug): I cannot access the web interface after some time unless I make a call first. The same applies for pinging the handset. It only will reply after call has been established. Might be a power save feature... :-) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Pertti Pikkarainen Gesendet: Donnerstag, 3. Juni 2004 08:26 An: [EMAIL PROTECTED] Betreff: Re: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails Prestige 2000W is the same BCM phone that was earlier referred as Wifi-600 in this list. http://www.bcm.com.tw/product/pdf/pdf1/Spec-WiFi600_2003_1103.pdf It has the same problem. If you enable WEP encryption ( 104 bit ), the voice becomes very choppy. Almost unusable. Without WEP it is fine. I wonder if anybody has better results with WEB enabled and with latest software releases ? -- Pertti Lars Boegild Thomsen wrote: I have noticed this one and I have also informed ZyXEL, but their response was vague to say the least. It is correct that the ZyXEL phone does not send a SIP Cancel when you disconnect an outgoing call that has not yet been picked up by the remote end. I have several times asked ZyXEL to put a formal bug report procedure in place with proper tracking but to no avail. Regards, Lars... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dominique Kull Sent: 02 June 2004 22:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ZyXEL Prestige 2000W SIP hangup fails Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying since it always leaves an empty VM. thanks Dominique ___ 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] ZyXEL Prestige 2000W SIP hangup fails
Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying since it always leaves an empty VM. thanks Dominique ___ 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
