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Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP

Hi, all This is certainly very good news! http://www.neowin.net/comments.php?id=21119&category=main

Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

Hi all, I have developed a full featured Windows IAX phone based on LIBIAX library . It is now in a prerelease version (0.9.0) and you can download it for free from my web page: http://www.laser.com/dante or http://www.geocities.com/tdanro Some of the features are: - registering with Asterisk PBX; - can use any audio device as ring device (including PC speaker), independent of the play device;

Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with "1" in order To successfully make a call to other USA destinations? ---- I have not been to USA (yet) :) Ta SJ

Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

Hi, I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks

Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? Ta SJ

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

...isk-users@lists.digium.com > > Is this bascially setting your bandwith value =3D high inside of = > iax.conf? > > Or is there another place to designate the codec? > > Thanks, > Wiley > =20 > > -----Original Message----- > From: Senad Jordanovic [mailto:senad@boltblue.com]=20 > Sent: Monday, July 19, 2004 2:11 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - > Advantages > > Yes, I don't have any bandwidth issues, so I could use 2 Mbps for 30 > calls. I do have issues with process...

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of empire underground Sent: Friday, January 09, 2004 1:32 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Screen Pop & Remote Agents > can I put a .csv file in the sql DB and have it dial from there? and will I be able to set a > Dial Plan to

...-Peter Junghanns) 9. Re: pri with TE410P not working (Austria) (Michael Bielicki) 10. RE: Cost of IP Phones, or Isn't It Just Software? (Andy Powell) 11. Re: pri with TE410P not working (Austria) (Peter Svensson) --__--__-- Message: 1 From: "Senad Jordanovic" <senad@boltblue.com> To: <asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] Soekris Engineering net4801 Date: Thu, 17 Jun 2004 08:34:01 +0100 Reply-To: asterisk-users@lists.digium.com John Bittner wrote: > Hi, > > I have it working great. I have debian running on it with music on &...

Hi, I am trying to figure out if * can register as a client on a remote MGCP service. Just like SIP and other protocols Do. Anyone tried this? Ta SJ

Hi, I am trying to get priority + 101 to work with Dial application. My dial plan is like this: [dial-mobile-peak] exten => s,1,AbsoluteTimeout(${ABSOLUTETIMEOUT}) exten => s,2,Dial(${TRUNKONE}${CALLEDNO:1}) exten => s,103,AbsoluteTimeout(${ABSOLUTETIMEOUT}) exten => s,104,Dial(${TRUNKTWO}${CALLEDNO:1}) I have changed password for first trunk to simulate trunk failure. Trunk one

I am sure that quite a lot of people would like to have Caller ID working with their X100P and TDM400P cards outside of USA. Judging from previous threads this is just a matter of implementing this support in the software driver! So, I was thinking, if we get together and put few $(USA DOLLARS) into a basket, we could then ask Digium to actually properly implement Caller ID for non USA

Does anyone know if SetGroup and CheckGroup apply to only current context or is it per server based? Ta SJ

Hi, Anyone know is there a limitation on internal asterisk database size? Ta SJ

asterisk-dev-bounces@lists.digium.com wrote: > I have a DSP based system that is working on a four port FXS system > using a 200MHz arm processor. Well.. since we are talking about this topic I owe you guys notes of my experience with SC1100 CPU used by various boards (www.soekris.com , www.pcengines.ch etc.). We made a Linux distro and compacted it into 32MB flash. Installed asterisk and