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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?

Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem

has anyone seen this? -- Starting Skinny session from 192.168.13.102 -- Starting Skinny session from 192.168.13.102 triton*CLI> Oct 15 13:44:05 WARNING[213019]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected. Oct 15 13:44:05 NOTICE[213019]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Success Oct 15 13:44:05

Hi, Is there anybody in this list who had experience with two TE410 cards on a server ? I know that the cards can?t share IRQs and I?m seeing to have two cards on a x335 IBM Xeon server. TIA -- Juanjo sin .sig

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

As per my problem yesterday with the Cisco 7960 and getting it talking to Asterisk on a different subnet, I gave up trying and just put the Asterisk box back on the internal subnet. However, I made two changes: - the external IP address is set on an ethernet alias eth0:0 - the main Linux router will change outgoing requests from 10.1.1.2 to the external IP (rather than the default behaviour of

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. --

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

Owias, The 537 is, for the most part, a drop-in replacement for the Digium card. Please search the archive and the Wiki, as there have recently been several discussions about this exact subject. To my knowledge that is about the only modem that works, but keep in mind it is _not_ supported. That being said, follow the instructions on Digium's site for installing the X100P and you should be

Hi, I?m playing with the CVS head time limiting at Dial application, it just works fine but the only problem is that the caller isn?t hearing the warning message. I?m using a Cisco 7960 as the caller and a Polycom 500 as the callee. The audio is passing through Asterisk: -- Executing Dial("SIP/8992-9712", "SIP/8988|20|L(10000:2000)") in new stack -- Limit Data: --

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