similar to: MGCP IADs

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

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?

We have a Dlink DVG-1120M and were surprised that it was able to handle 2 simultaneous conversations to 2 seperate phones using only 1 MAC address and 1 IP address. So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in sip.conf to add a second line to a device. Is this possible? Can this only

I've never set up an mgcp device before. I have an Adtran IAD with the MGCP firmware on it. I have it configured in mgcp.conf like this: [general] port = 2427 bindaddr = 0.0.0.0 [adtran] host = 192.168.2.2 context = default canreinvite = no line => aaln/1 line => aaln/2 The device is configured like this: MGCP Configuration | Standard MGCP 0.1 / NCS 1.0 MGCP Endpoint

I am have trouble getting caller id to work here is my current mgcp.conf am I missing something? ========================================================= Mgcp.conf [general] port=2427 [Egraph-1] ;dlink 104s-1 host = 12.151.207.2 context = local line => aaln/1 callerid = "jim's office 1" <321> line => aaln/2 callerid = "jim's office 2" <322>

Hello, Anybody has it working with asterisk? Could you share your experience ( good/bad) Thank you -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

I'm considering switching my incoming phones lines from standard analog to a T-1 service from XO communications. They propose to bring in an "IAD" which has 12 lines of voice and 768k of internet bandwidth as part of a package deal. Since I want to keep the voice traffic in the digital domain the equipment they're proposing is a "Lucent Digital Vina Integrator" IAD

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

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

Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using

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

Hi all, I have VOIP network built with MGCP endpoints.The manufacturer of endpoints is ASKEY. I downloaded latest Asterisk software and found it very useful for me. I configured it and it seems taht everything works OK when I am testing it with one or two endpoints. After that I tried to move Asterisk to working network and replace existing call manager. It starts working and calls are

Hello, I`m try to make the attended transfer work Dlink DG-104S via FLASH, when somebody calls my phone I pickup and press flash to get a second line to call another extension. When I press flash I hear no dialtone, and only a long and then small beep. When I try to dial digits I hear again those long+short beeps, but the extension dialed is not ringing. If I pres flash again I get back to

Hi, After spending some time trying to get a DG-104S working behind NAT, I finally found the problem. I made the incorrect assumption that nat=yes in mgcp.conf works just like sip.conf. The channels within a gateway are treated more closely to zap channels than sip channels (from a .conf standpoint). What this means is that you have to put nat=yes BEFORE any subchannel definitions: This

As a follow-up to my previous post where I stated the IAD would be a Vina model, after some more prodding to XO, they have told me it will either be an Adtran TA-600 or a CAC Adit 600. These products are covered pretty well on the web and I have manuals on both. So, if those knowledgeable folks had to use one of these to attach to an Asterisk box, what interface would be best or at least

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

I have been reading asterisks and everything I can get my hands on for the past week. I want to know what class processor is the bare minimum I need for a four port Asterisk installation? Thanks

Hi all, A new version of DIAX (0.9.8a) is ready to be downloaded from the following locations: http://www.laser.com/dante or http://www.geocities.com/tdanro What's new in 0.9.8a: - unconditional autoanswer or based on CallerID (user configurable); - use any Ericsson/SonyEricsson GSM/PCS to control DIAX (feedback on the phone display) through Bluetooth (or serial cable). You do not even

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 a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use 24 analog phones with it however something really weird happens I can place calls from my ata,