similar to: GSM 6.10 codec for Asterisk

Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server connected to Twinkle and X-Lite clients. I have to use the GSM codec for all of my clients, and it was set up in the sip.conf specifically in "allow=gsm" line. Twinkle has GSM codec built in, but when I open X-Lite audio codecs settings I can't see the GSM codec, being that the official web site and the PDF

Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented with GSM sound files. The problem is I have IP phones Utopix HyperPhone 202 which support only G.729a/u and G.723.1 high/low, but not GSM. If I choose G.729A the "pass-throu" calls among users are OK, but Asterisk can't transcode GSM to G.729A in voicemail calls. My company doesn'y want to pay for a G.729

Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all, I have Asterisk 1.4.13 and I need to use encryption among Asterisk and my SIP users, and with the RTP data interchanged among users. I prefer the use of ZRTP/SRTP because we use Twinkle and X-Lite/Zfone as our voip clients and they support these encryption mechanism. My question is: do I have to enable any encryption support in Asterisk 1.4.13 ??? Or Asterisk has native encryption

Dear all, I have implemented an Asterisk SIP server for a WAN VPN over Internet. We have users distributed along all my country (Argentina) that register to my Asterisk in order to talk among them. I'll plan to have voice and voicemail with GSM codec, because we can't afford the payment for the G.729 licenses (it's an administrative problem of our company, not an echonomical problem).

hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next

Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The SIP clients are using different operating systems such Debian, Gentoo and Windows XP so they use different SIP softphones like SJPhone, Twinkle and X-Lite. In order to let SIP clients to see the presence status to each other, do I have to establish any special setting in Asterisk 1.4 ??? Or the presence status (online,

Dear, I need to create a custom device extension in order to dial a local queue. Suppose my queue number is 8888, how can fill the Dial field from the custom extension ??? Because if I put just 8888 or Local/8888, I don't succeed. Thanks a lot

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP with Linux/Debian Etch??? I'd like to see if my intranet contacts are available, busy, disconnected.... Thanks a lot Alejandro

Dear all, I have to install an E1 card in my Asterisk 1.4.23 server and here is my short question: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card ???? Thanks a lot Alejandro

Hello, I have an error while try to install twinkle: # yum install twinkle [...] Resolving Dependencies --> Running transaction check ---> Package twinkle.i386 0:1.2-1.el5.rf set to be updated --> Processing Dependency: libresolv.so.2(GLIBC_PRIVATE) for package: twinkle --> Finished Dependency Resolution Error: Missing Dependency: libresolv.so.2(GLIBC_PRIVATE) is needed by package

The Gnu Telephony site: http://wiki.gnutelephony.org Has a Centos repo: http://dist.gnutelephony.org/RPMS/ But I caught some text stating that this is for Centos 4.2. Is it really? Is there a difference; i.e. would it be safe to install these on Centos 4.4? Really I am after Twinkle, and it seems there is a lot you need to actually get Twinkle installed...

*I have a Ham Radio program, named CQ100, it works fine using WINDOZE, but when I installed the same program on my linux system everything works except there is no-audio I'am using Ubuntu 7.10 linux... The author told me that windoze uses GSM 6.10 codec, plus ACM audio compression manager, these are built-in... So by anychance does anyone know of a program that one can get to use on a

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to test with a friend who has an Asterisk in his office with the Asterisk which I have in my house. Then I have an extension that he is trying to register remotely. Trying with the Twinkle client, I see that it is registered: - --------------------------------------------------------------------------- 400/400

Dear all, I need to build a PBX based on Asterisk for a call center. I have worked with raw Asterisk but it's hard to work for big implementations think. Also I have worked with Trixbox CE for a small bussines and it was prette good, but I have not have many features like ACD. I know there is another version called Trixbox PRO -specially Call Center edition- that's not free but has got

Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with four HD's available, using CentOS as the OS. What's the best RAID type recommendation ??? RAID 1 or RAID 5 ??? Regards Alejandro

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

I have Asterisk 1.2.13 installed as a Debian package and I edit only sip.conf and extensions.conf in this way: sip.conf: [general] realm=work.com.ar ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes