similar to: Ring delay 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 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 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 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 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

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, 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

Dear, I have an Asterisk PBX with two E1 cards: Digium TE122 and Sangoma A101D. Sangoma card has SNMP support but Digium card has not, and also SNMP does't give me ral time information. Within CLI Asterisk I execute "dahdi show channels" but I don't get information about channels usage. What is the best way to have real time monitoring of E1 channels usage and status ???

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'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 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 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, 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, I always had one E1 card with one span, so I've never had any problem in set it up through /etc/dahdi/sustem.conf and /etc/asterisk/chan_dahdi.conf because I put span=1. But now I have a PBX with two E1 cards with 4 span (8 span in total). How do I have to define both card in system.conf and chan_dahdi.conf, and how do I have to refer each span to the corresponding card ??? Thanks a

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

Dear, I have this scenario: - PBX Asterisk 1.6.2.10 with private IP 192.168.0.10 - Behind a Cisco ASA firewall that connects to Internet - SIP trunk to Net2Phone with these parameters (nat=no): host=200.58.113.60 username=DOLLY secret=123456 port=5060 type=peer dtmfmode=rfc2833 disallow=all allow=alaw&ulaw nat=no canreinvite=no qualify=yes -Softphones Xlite The PBX can't register to

Hi, I've been meaning to fix my non-working paging feature here for a while, and I've just spent the last 5 hours looking at many, many web pages that all say the same thing. I am using Asterisk 1.6.2.18 and Polycom phones, both older (501 with "latest" legacy 3.1.7 firmware) and newer (335 and 650 with latest 3.3.1f). I have changed the correct values in sip.cfg like

I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the