similar to: Asterisk and G.729 codec: short questions

Dear all, I have an Asterisk SIP PBX using GSM codec at peers and in voicemail sounds files (I have Spanish sounds). But now I have a problem because I have to use G.729 mandatory at peers, and I have GSM in voicemail sound files. I can't let Asterisk do trascoding because I have no a DSP in the CPU, and I don't want to degrade the PBX performance with trascoding tasks. So how can I

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've read that Asterisk supports only the G.729 A audio codec. I have several Grandstream IP phones with G.729 A/B codec implementation. Does G.729 A/B mean both version A and version B, or A/B is a new version different from A and B and it's not supported by Asterisk ??? Thanks a lot Alejandro

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 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).

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

Just a short question: I will have Asterisk using G.729 codec and connected to some voip devices such IP phones (GarndStream) and a GSM gateway (Portech). Do IP phones and GSM gateway include valid G.729 licenses or do I have to pay for them ??? Thanks a lot Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

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, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both boxes. I need to connect both PBXs with E1/R2 and UTP cable. What are the requirements to deploy the UTP cable ??? Straight-through or crossover ??? What are the pinouts in both peers ??? Thanks a lot, Alejandro

People, I will install asterisk on my Debian Etch box without a PSTN interface card. I want to use only softphones for the moment. My question are: 1) Is it enough to install with "apt-get" the asterisk 1.2 or do I have to get asterisk 1.4 manually ??? 2) Do I have to configure a dummy PSTN interface in my case ?? And if you have a debian-asterisk howto, I really thank you. Regards,

Dear, when I dial *30 in order to get instructions to blacklist an extension, Idon't get the menu but I get a new dial tone. What happen please ??? What can I do to solve this ??? Thanks a lot, Alejandro

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 use Trixbox as my PBX. Until a couple of days I've installed a GSM Gateway to communicate with our three cellular phones: 15 64227777 15 64228888 15 64229999 The GSM Gateway has just one SIM. I use the Free PBX web interface in order to set up the route and trunk parameters: Trunk: ******* Name: SIM1 Peer details: host=10.10.1.2 (IP from GSM Gateway) port=5060 type=peer

Dear all, I've read some news about Sisky (http://www.yeastar.com/Products/SiSkyEE.asp), a service to interconnect Skype clients with SIP clients. Does anybody test Sisky and can tell me about his experience ??? (Sisky runs on Windows because Skype and its API are more stable on this OS). Regards, Alejandro

Dear all, I have Asterisk managed by a FreePBX web console, and I want to add an external dial code, in order to dial 9 to get external line/tone for outgoing calls to the GSM network through my GSM gateway. Where from Asterisk/FreePBX can I setup this feature ??? Thanks a lot. Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all, I need a call center asterisk's based solution and I see there are two important solution for 120+ agents: VicidialNow and ViciBox Can you tell me the difference between these open source call center solution please ??? Special thanks Alejandro

Dear all, I'm planning to buy some IP phones with GSM audio codec support in order to use with an Asterisk SIP server I have implemented and nowsuccessfully running with softphones like Eyebeam and Twinkle. A vendor offer to me the SNOM 300 IP phone, that support GSM 6.10 audio codec. I've looking for GSM 6.10 codec in the web but there is no helpful information. Just I enter the