similar to: SIP Dial Command to a non-Asterisk url

set_format: Unable to find a codec translation path from ulaw to g729 Both endpoints are PAP2 set to G711 only I have 1.2.17 on Suse 10.1

Hi: I wonder if anyone can help me about cpos function not found error: : path.package("cwhmisc", quiet = FALSE) [1] "C:/Users/slee/Documents/R/win-library/2.15/cwhmisc" So I have package cwhmisc where there is cpos function. But I got error: cpos("ab","b",1) Error: could not find function "cpos" Then I tried to install on R prompt but got this

Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue

hi! First of all i want to tell i have a dedicated server on layeredtech with direct internet connection and i currently dont use iptables, so this is not about network configuration =). well so, i install asterisk-1.4.2 on my server, and next install asterisk-gui from the digium repository. next i go to: http://pbxa.com:8088/asterisk/static/config/cfgbasic.html and install a default

Hello eveybody, I've been looking for a way to share trunks between two asterisk servers. I guest I have to use Dundi, but I've not found the exact method yet. I need a way to allow users registered in one server to use the another server's trunks in the case the first server's trunks were busy and vice versa. Is this possible? Thank you so much, any comment will be useful.

Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip

Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? I use Asterisk now for my phone system. Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:wsiler@education2020.com <mailto:wsiler@education2020.com>

Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html But it will be 3 PCI slots. Thanks, Gavin.

Hello. I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. One feature the SPA-841 has, which I can't figure out how to implement on the snom 190, is the "make/accept calls without registration" feature. Or more specifically, "produce a dial tone even if I'm not registered." I would like to set our

Could you kick me, I can't dial more then 9 digits. Is anyone some default length of extensions or dialed number. Thanks, Bob.

Dear all, Could this be added to http://dovecot.org/tools/ http://cpan.org/authors/id/G/GH/GHENRY/create_dovecot_shares-1.05 Thanks. -- Kind Regards, Gavin Henry. Managing Director. T +44 (0) 1224 279484 M +44 (0) 7930 323266 F +44 (0) 1224 824887 E ghenry at suretecsystems.com Open Source. Open Solutions(tm). http://www.suretecsystems.com/

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

Hello, I'd like to implement some public sip uri's that poeple can call into and get an echo test. Is there a way to force a codec so that users can test various codecs? Something like: echo-test at example.com (negotiates whatever codec, is there a way to figure out what codec was negotiated and tell the user) echo-test-g711 at example.com (forces g711) echo-test-g729 at

Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming

We will be connecting our Asterisk server to ISDN 30 and intend using the Sangoma A101 card. The install location is in London (UK). Sangoma card at Voipon http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA I would be grateful to hear if this is the right choice of card. Usage reports would be helpful. Regards Rory -- Rory Campbell-Lange

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

Would anyone have any ideas with smbclient sits at FUTEX_WAIT in below message? Typing: strace smbclient -d 5 shows it sitting at: futex(0x2aaaabdf2dc0, FUTEX_WAIT, 2, NU. SUSE 9.3 x86_64, Samba RPMS from main Samba site rpm -q samba-client samba-client-3.0.23a-0.1.34 -- Kind Regards, Gavin Henry. Managing Director. T +44 (0) 1224 279484 M +44 (0) 7930 323266 F +44 (0) 1224 824887 E

I was just curious to what your redundancy solution is. I have considered many options, so I thought I would share and get an idea for what others are doing. My setup is two different locations with a 10MB WLAN fiber link between the two. Each location has it's own PRI as well. I have considered and tested many options this last year or so. 1) Using hearbeat and drbd to monitor the

Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,

Dear all, Getting lots of these errors in our maillog with the INDEX changed: dovecot: IMAP(ghenry): fchown() failed with file /home/ghenry/Maildir/index/.INBOX/dovecot.index.tmp: Operation not permitted For .log and .cache as well. We have: location = maildir:/home/%u/Maildir:CONTROL=%h/Maildir/control:INDEX=%h/Maildir/index I looked in safe-mkdir.c and saw: fchown() failed for (line 42)