similar to: Cisco AT-18x SIP 3.0 Firmware

Thanks, Mark! Here's a summary of what one needs to do in order to get NAT working with Asterisk. Please note that I have a Cisco ATA-186, and your experience may be slightly different based on the equipment you're using. You'll need to have a CVS updated version of Asterisk as 2003-03-06 ~2:00 PM EST. NOTE: This currently works for outbound calling only, not inbound. In other

A friend pointed out this url http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists intercom/auto-answer as being a feature in Cisco Call Manager (which as I understand it, uses SIP predominately for handsets). I've come across comment somewhere that intercom isn't supported in the SIP spec. Does anyone know if the apparent capability of Intercom being available in SIP

I have a cisco 7940 with the following sip.conf config: [Desk1.1] type=friend secret=****** defaultip=192.168.1.14 insecure=no mailbox=102 callerid="Desk1.1" qualify=500 canreinvite=no context=extensions host=dynamic group=2 I do not get message waiting indicator (mwi) on this phone. Is the another .conf file invilved in configuring this function other than the mailbox=xxx in the

Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

Hi, An Cisco 7960 ipphone has been set to SCCP firmware by one of our students. I want to set it to 7.5 SIP firmware and I've been unsuccessful yet. Firmware versions are SCCP 3.0 (Source: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1045789) ie: Application Load P003F300 Boot Load ID PC030300 When I browse, phone settings, I see

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -----(H323/SIP)------> Cisco ----- (E1/PRI)------->Telco So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go out H323 or SIP to Cisco, where they go out PRI. I have the Asterisk side sorted :) (either H323 or SIP), I need help in the

Hi, For MGCP users. Is there any success stories with any MGCP IAD vendor. I?m trying to find an IAD which works with Asterisk. I?ve tried the Cisco IAD 2430 without success; but SIP on this IAD works but it?s limited (no authentication, no notify messages, etc) and with higher density IAD (16 or more ports) it?s nice to control using MGCP. Any information will be apreciated ! Thanks. --

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateFormat = D-M-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 languaje=es allow = all ; disallow

I started thinking that perhaps I should move my TODO list to a bug tracker. But because of reasons I've explained a few times before, I don't want a full blown public bug tracking system. The requirements for it are: * I am the only person who can add new bugs. Everyone else reports bugs/requests to this mailing list as before. (Well, I guess Stephan could use this as well if he wants

Dear friends, I have compiled and installed h.323 in my asterisk. And I have a service from a H.323 VoIP provider who give me user, password and gatekeeper IP address. All configured. But when I start my asterisk i receive the following error and h.323 calls can not be making and/or receiving. [chan_h323.so]=> (The NuFone Network's Open H.323 Channel Driver) == Parsing

Should the following setup work? SIP UA---NAT---Internet---NAT---SIP UA If both UA's support STUN and report the external IP address in the SIP packet.. I am trying to get away from using canreinvite=no so that traffic can go directly between the UA's and not via the central server but I can't seem to get it to work.. Has anyone set this up and can give me some pointers??

A file is chmod 600. It gets opened on Windows, and it gets changed to 644. Now the secrets are exposed to all users. This is bad! This happens if the user does: N:\> echo foobar > secretfile But it does stay at chmod 600 if he does (append instead of truncate): N:\> echo foobar >> secretfile Why does this happen? The "create mask" parameter is set to 644. I do not

Hi, I''m trying to make an app to control a set of machines, and I need to executes linux commands for it. I''ve been looking for in Google... How can I execute system commands with Rails and control the commands output??? How can I find a good tutorial about threads??? Thanks, Juanjo -- Juan Jos? Vidal Agust?n <juanjova@um.es> ATICA - Universidad de Murcia

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

Dear Sirs, I'm running an Asterisk 0.9.1 in a Fedora Core 2 box. I installed a X100P card on my box and when i try to load modules i am rejected. When i type modprobe zaptel my Fedora respond : "FATAL: Module zaptel not found." . The same uccurs when i type "modprobe wcfxo" May you help me. Thank you in advance Juanjo

Hi, I?m following the Depot tutorial in Agile Web Development with Rails. I need some help to display products thats in one specific category. mysql> select * from products where category = "jackets" does what I want, but I?m not sure how to make a list and display it in rails? This code displays every product with some of its fields, but I want it to only display the products

I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem?

Hello, I have finally received the details from Polycom to get into the backend configuration of their SoundPoint IP 600 SIP VOIP phone. The phone is quite nice looking but the configs are very sparse, not even a place for a secret(password) field in their SIP registration section. If anyone else has one of these and needs the passwords to get into the back end configurations, just send me an