similar to: unknown RTP codec 126 ??

i am asked to implement a call center of 50 seats for my company , and i was wondering if Asterisk can fit this as a relaibale and low price system is it mature enough for this task?? best regards Gers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090712/b5bab36d/attachment.htm

does Asterisk suppoet CSTA protocol for CTI applications? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090722/a2149cc5/attachment.htm

does anyone use OpenSBC , or know if it is mature stable opensource for a production enviroment http://rpm.pbone.net/index.php3/stat/4/idpl/10795970/com/opensbc-1.1.4-3.el5.pp.i386.rpm.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091201/67c95aac/attachment.htm

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

Does anybody know if i can make (chan_mobile) module to be installed and work with Asterisk 1.4.19 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080513/57510b8a/attachment.htm

can i use MP3 files as an IVR prompts directly without converting to .gsm format? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091010/67205e07/attachment.htm

Is there a way to ensure that the source IP address from witch the SIP user register is not tampred with , is there a feild in the SIP register message header can be used to achive this security ? i have an asterisk server in witch SIP users register through an SBC(session border controller) , i wanna make sure that those users are really registering from the IP they are claimming they are

Hi all, I want to use a cell phone as my FXO line to Asterisk Box ,did anyone try this and configured it and how to physically connect it to Asterisk server? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080506/6fafcfba/attachment.htm

Thank you, I needed a starting point to start my post. 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. Voice issues on IAX2 Trunks, All extensions are SIP. The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 set debug trunk on [2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793 compress_subclass: Can't compress subclass 2097217 On the box running

Hi! I use Bria and eyebeam and it seems that asterisk doesn't send RCTP keepalives when a SIP channel is on hold. This is a known issue as is described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Bria This gets very annoying because very often people are put on hold longer than 30 seconds (the phone's default.) In a company with more than 100 soft phones

Hi, I would be curious to play with an Android phone with Wifi-only capability. My plan is to install Bria on it and see if it could be used within a couple of WiFi access points, as a high-end wireless phone. Which handset would you recommend ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is there any softphone or TAPI plug-in that allows one to dial from a web page? As you may know, Skype has a mechanism that converts phone numbers on a web page to a click-to-dial application. I'd like to use this but on a normal softphone (Bria, Xlite, other). Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear, Kindly guide with the 2 issues mentioned below *#1* - *Host unreachable 0 last qualify 0 (only in 3G**)* I am trying to use SIP client over 3G. It registers and call can be initiated from the client but it can't receive call; cause *asterisk sever *marks it as unreachable immediately after registration. "[2014-10-08 14:32:47] NOTICE[1610]: chan_sip.c:29596 sip_poke_noanswer:

i am trying to make a video call on asterisk 1.6 , my configuration is an - asterisk 1.6 on Centos on virtual machine VmWare - Xlite softphone one windows xp (the Host operating system) - X-lite client on another windows XP (the Guest operating system ) i put the paramter videosupport=yes under the general section in sip.conf i allowed the video codecs for each client in sip.conf for

Hello, I'm trying to use presence with PJSIP and I have a "issue". I created correctly hint priorities like: exten => 1000,hint,PJSIP/1000 exten => 1001,hint,PJSIP/1001 Extension 1000 can subscribe extension 1001 y vice-versa. The problem is when the extension 1000 make or receive a call. In the softphone where the extension is present on buddy list, the extension appear

Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

please can anyone help me knowing if i can install Linux and Asterisk on HP servers _________________________________________________________________ Put your friends on the big screen with Windows Vista? + Windows Live?. http://www.microsoft.com/windows/shop/specialoffers.mspx?ocid=TXT_TAGLM_CPC_MediaCtr_bigscreen_012008 -------------- next part -------------- An HTML attachment was

Hello, I am using asterisk 14.2 and PJSIP, with TLS transport. I?m sure I?m doing something wrong here .. In 2 distinct softphone clients (Bria and Groundwire), I am able to register successfully, and place a SIP call, with no certificate warnings. But shortly after I place that first call and hang up, I receive a certificate name mismatch error in the softphone, the error presenting me

Hi all, my name is Arnau Bria and I''m a sys admin in a center where we must deal with hundred hosts. We''re currently working with quattor, but it''s too complex for our purposes, so I''m looking for new admin tool. I''ve been playing with CFengine for few days (2 or 3) and I''ve seen some limitations that makes me thing that CFE is not our

Hello, Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone. Is there an better softphone? Or are there softphone solutions for PC desktop MAC or Android with an headset? I want to save cost for desktop phones. thanks Thomas