similar to: asterisk setup w/ voIP phones

Hi, We have zabbix running and would love to be able to monitor our asterisk box with it. I believe that some sort of SNMP is build in 1.4+ correct? Where do I find more info or a how to on what is supported and how to use it? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal

I know I can setup asterisk without Internet at all and it works as local pbx. Would an asterisk box work with a dynamic IP, with a dyndns name? What must I take care if I try that? bye Ronald

Hi, My apologies for the OT. My googling came up empty and hopefully there are some members in the community that could give me a hint how to solve this issue: Cisco 7961 with SIP firmware 8.3.3. Needed to downgrade it to 8.0.4SRS2. The downgrade process started off good. The 7961 got it's IP address via DHCP, found it's SEP<mac>.cnf.xml file and started to "upgrade"

Hi there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat

Hi folks, Am new to asterisk pbx systems. I am trying to figure out what to do, I'll list and folks feel free to give feedback and advice. MAIN purpose for usage: 1.exposure to setup an asterisk box 2.get home phone service via VOIP/internet connection. tasks so far ------------------ 1. setup and install asterisk (1.4.x) --> DONE -currently configuring sip.conf

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate

Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link: http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html Please feel free to comment on the

Version - Asterisk SVN-trunk-r12793M (1.2.4) I have 4 Grandstream GXP 2000 phones configured. However at the moment, I have had to disable BLF, Hints, and Call Limiting due to an extremely annoying bug which seems to make the phones channels "lock" in busy after a call has been hungup. If I do a show hints after say extension 200 has hung up I get the following -= Registered

Hi All, I have working asterisk version 1.4.24. I have a blind transfer issue with grandstream bt200. I have updated the latest firmware to the phone. The phone is sending the *refer* to asterisk but asterisk is not able to connect with the another call that i have checked in sip debug. I am using transfer button of the grandstream phone. Can anybody provide help for this issue? Thanks in

I''m a total rails newbie, but I''m learning fast. I have a question that I can''t seem to find an answer for: What is the best way of coding views that represent foreign key relationships? For example, while developing, I''ve created an "articles" table. I create some scaffolding code and modify it all to look nice. I then realise, I need an

Good evening. I have this following table and I would like to turn it into a matrix in which my rows would be filled with de "Sellers", my columns with my "Articles" and my data would be the mean unitary price used by each seller in each produt. Seller Art. Unit Price 1 v1 p1 9.148352 2 v2 p1 2.858073 3 v3 p1 3.775315 4 v4

Is there an ATA type device out there that has low level audio out for connecting speakers? My asterisk server is in one building, I wish to have speakers in another building and connect them up to a low level audio device that I can call into and speak. Can I connect speakers into the FXS or FXO of a grandstream HT 503? Does that work? I'd prefer not to have a PC setting there (sound

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing with a Grandstream BT200 telephone and, according to I read, it has a LED that blinks if for that extension messages were left. In "Voice Mail UserID", under the "ACCOUNT" tab, I put *100 that is the extension in which my Asterisk answer the voicemail service and if then I press MESSAGE button, the

Hello list, I have an app that has a very simple authorization scheme. A person can have many roles and roles can have many people. In my app, I''d like to do before_filter :login_required (since no role name is provided, it accepts any users with credentials) before_filter :login_required ("administrator") (only accepts those with role administrator) before_filter

Hi - OK not really an Asterisk question but it is affecting one of my favorite features - emailing voice mail! I've posted on some Linux forums and sendmail.org but no response so I'm hoping someone will take pity on me ;-) My ISP requires SMTP authorization and I'm having a heck of a time getting it to work. I've included the following below: Asterisk 1.4.21 CentOS 5 Sendmail

I have an B2B application where a pretty complex order form needs to be submitted and edited on the admin controller, buyer controller and seller controller with some small differences. How do I make available the edit order methods all controllers? -- Posted via http://www.ruby-forum.com/. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are

What is the problem with SIP retransmits? ----------------------------------------- Sometimes you get messages in the console like these: - "retrans_pkt: Hanging up call XX77yy - no reply to our critical packet." - "retrans_pkt: Cancelling retransmit of OPTIONs" The SIP protocol is based on requests and replies. Both sides send requests and wait for replies.