similar to: adress book

Hello, When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway, there is problem with DTMF "out-of-band". See debug below: Mediatrix forces (*) to use Payload Type as 96: [...] a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [...] Then we've got this nice debug from (*): May

Dear All : I need to link my HQ to some Remote Sites - I need a Router which supports VOIP , and VPN Also the Router Should has 3 FXS ports and 1 FXO ... The call should be routed from the Remote Site to the HQ through a VPN tunnel ( 3DES ) ... Any Advise ? Mohamed Farid ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are

Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we

Hi all I have a pocketPC Toshiba e750 and I want to make SIP calls from it, but I didnt found any free softphones for my Toshiba. X lite's versions for pocketPC isnt free :( Did someone used before a free softphone for pocketPC? witch one? Thanks Joao Pereira www.fccn.pt

Hi, How often do you all have your ATAs and phone register with the asterisk server. I am doing it once an hour, but now I am wondering if maybe that is too long in between registrations?

All, does anyone use the Oasis DSML standard? It''s a mapping between standard LDAP/LDIF operations and data structures to XML. There are some Java tools that handle DSML. I just saw a requirement for accessing an LDAP server by a service that requires results encoded in DSML, so I thought I''d better add this support to the Net::LDAP library. Is this of interest to any of

Hi, We are currently testing Asterisk with Wildcard 100XP and serveral Cisco ATA Box. Everything works great except that the card does not detect the hangup signal. We are using a standard Belgian PSTN line. I have not found anything about a be zone (only us, fr, de, nl, ...). Does someone experience the same problem? Do I need to create a new zone be (and how to do that)? Another small

Hi I tried Xten, its very good, because it can stay in the taskbar (next to the clock) and start when windows starts, and is allways ready to receive calls. Maybe it s the best way to introduce VoIP to my company workers.... But theres a feature that s missing (or I couldnt find), there s no way to connect this softphone with the adress book. I think this feature is very important, because

Hi, We are using Asterisk running on FreeBSD, as IVR / Voicemail for SER. We have redirected certain calls from SER to *. On * there is some 'testing' extension. It's simply playing some demo now ;-) As long as I use plain G.711 the sound is nice. When I switch to G.729 the sound is choppy, not recognizable. What is going on? Debug shows everything is normal.. I understand that all

I've had a few problems with my system holding the line after a call has been made, just now I rebooted and noticed the following in /var/log/messages Aug 13 17:23:15 Sheriff kernel: wcfxo: DAA mode is 'FCC' in the file wcfxo.c the following structure is initialised as below which would suggest that FCC is wrong for France or pretty well all of Europe. static struct fxo_mode {

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

I had a similiar problem, but with calls dropping after 30 sec. It turned out that Android didn't support RP-CID (Reverse Party Caller ID) so when I sent the name of the callee to the caller (as some sort of "centralized phonebook function") it caused calls to be dropped as android refused to reply on the packets or sent rejections back. Check if you have some equipment on the line

Hello, I have Snom, Cisco, Grandstream & YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101202/ea25cd7e/attachment.htm

Hello, sorry for the bulk question, but I cannot find answer anywhere, I have model like Phonebook::Category, and i`m stuck on calling find() method on this model ?

Hi, I wanted to let you know that I have just made available the sources to the ogg + vorbis implementation in haXe, which I've been working on for last couple of weeks. The code compiles to an swf file playable in Flash Player 10. A demo of a simple player implementation (latest Flash 10 required): http://people.xiph.org/~arek/pg/hx/test.html and the sources, in a bzr branch, currently

Hi, I wanted to let you know that I have just made available the sources to the ogg + vorbis implementation in haXe, which I've been working on for last couple of weeks. The code compiles to an swf file playable in Flash Player 10. A demo of a simple player implementation (latest Flash 10 required): http://people.xiph.org/~arek/pg/hx/test.html and the sources, in a bzr branch, currently

Hi all, My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote an AGI script that does the following: 1) If it's a toll free number (800|888|877|866), set the CallerID name to "TollFree Caller" 2) Use curl to look up the number in Google phonebook 3) If a business listing, set the CallerID name to business name, as is. 4) If it's a residential

On Sun, 23 Aug 2020 at 18:23, Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On Saturday 22 August 2020 at 22:51:48, Sebastian Nielsen wrote: > > > I had a similiar problem, but with calls dropping after 30 sec. > > It turned out that Android didn't support RP-CID (Reverse Party Caller > ID) > > so when I sent the name of the callee to the

Dear all, As the next version of XiphQT is mostly ready, I thought it could use some more wide pre-release testing. The major change since last release is the addition of Ogg exporter and Vorbis and Theora encoders. Any feedback on how this new functionality performs (or doesn't!) with audio/video editing/producing software will really help. Also, comments and suggestions on the work of