similar to: can't build resODBC on SUSE 11.3

I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor

This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of

I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card. Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable? If so, any clues where I might buy one in the UK? The Digium card sellers don't seem to stock such a thing. Thanks. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk

After working fine for a week or so my new Quad E1 asterisk 1.8 system has started rejecting outbound calls from the Nortel BMC 450 it is connected to. The cli fills up with these: sig_pri.c: Ring requested on unconfigured channel 255/255 span 3 Is this likely to be a 1) config error 2) cable issue (I made them) 3) hardware problem with the Digium card 4) software (lib pri) Any clues?

> Asterisk Project Security Advisory - ASA-2007-013 > > +----------------------------------------------------------------------------------+ > | Product | Asterisk | > |----------------------+-----------------------------------------------------------| > | Summary | IAX2

> Asterisk Project Security Advisory - ASA-2007-013 > > +----------------------------------------------------------------------------------+ > | Product | Asterisk | > |----------------------+-----------------------------------------------------------| > | Summary | IAX2

Dear Tim; What is folks? Where I can find it about VPN solution? Regards Bilal > Hi, > > Greetings to All, > > Im looking for some help on configuring VPN on the Asterisk PBX that I > have hosted in US. Im currently in Middle East and as everyone knows > some countries here has taboo to VOIP. Im not able to get phy phones > registered to my PBX as they are blocking SIP

On 8 Oct 2010, at 15:37, Danny Nicholas wrote: > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim Panton > Sent: Friday, October 08, 2010 9:34 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Weird stalling of playback on IAX2 channels

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

Hi, I am using a 64-bit RHEL 5 machine. I built Asterisk latest 1.6 branch. The system has ODBC and Postgres installed. psql, isql and odbc work fine. Asterisk "make menuselect" for some reason does not see the installed packages and refuses to build res_odbc and other packages. How do I force it to do that? Is there a way to modify the output file from menuselect and make it

Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these

Hi All, Thank's for your hint Tim Panton I could connect my asterisk machine to my oracle machine. I used unixODBC-2.2.11.tar.gz, oracle-instantclient-basic-10.2.0.3-1.i386.rpm, oracle-instantclient-sqlplus-10.2.0.3-1.i386.rpm and the drive from www.oracle.com (odbc-oracle-3.1.0-linux-x86-glibc.tar) to configure my asterisk machine. I can connect to my oracle machine with isql and in

HEllo list ! When user A calls user B via Asterisk (Users A and B are registered on the same Asterisk server ) and an ISDN PRI, user B phone always shows Asterisk server telephone number. How to hide it and how to forward user A number ? We tried usecallerid, callerid, hidecallerid, restrictcid, usecallingpres in zapata.conf but we always see Asterisk server telephone number ! Thanks

Ooops did Google just become a carrier :) http://googleblog.blogspot.com/2007/07/all-aboard.html I hear stocks crumbling worldwide as I type. Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070703/92218fc6/attachment.htm

I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it

I'm having trouble getting a xorcom set up. A large part of the problem is that the box is a _long_ way away and I can't get to/at it easily, so while I could probably fix this in a few hours if the machine were in front of me, I'm struggling over a slow unreliable laggy link. Ok, enough whining from me. I have a new Xorcom plugged into the usb of a Suse 11 machine I built Dahdi

I think I've forgotten something obvious.... I've got 2 incoming calls, I want to bridge them - how can I do this ? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital.

I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason

Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon

hi, i'm searching solution for playing media(moh,prompts,voicemail,recordings - wav format) from adobe flash player (web browser) flash cannot play wav directly (imho) i must convert files to any other format on-the-fly - i cannot use mp3 because of royalties - next option is swf (with ffmpeg), supported free audio codecs (http://en.wikipedia.org/wiki/Flash_Video#Format_details) *