similar to: setting up asterisk as voicemail for softswitch

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its

I have an audiocodes mp-124C/FXS gateway that was a leftover from a failed voip experiment and I have no documentation at all for it. Anyone have a manual in pdf form they could share? -- Chad Whitten Network Administrator neXband Communications cwhitten@nexband.com 601-944-4801 Phone 601-944-4803 Fax

ive had enough of cisco unity and microsoft exchange and im looking for alternatives to our voip system. right now, we have 3 cisco callmanagers, 1 cisco ip icd system, and 1 cisco unity voicemail system. all phones are cisco 7940/7960's and some ata186/188's. voice gateways are cisco vg200's with pri cards (5 total). im running h323 on the gateways and phones are of course

Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that.

Has anyone had any success using the Adit 600 with the CMG card talking MGCP to asterisk? I want to have a central asterisk server with 10 Adit 600's at various locations providing 24 FXS ports.... Thanks, Isaac

Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router using MGCP IP protocol, instead of controlling it through an E1. Have anyone tried this configuration? How does MGCP works? I've tried to search for it on Google, but I only find the protocol specification for it. Is Asterisk fully capable of this? I can't find any documentatin covering the use of MGCP in Asterisk.

I have been searching for some documentation that would indicate if Asterisk supports H.248 and everything I have come across seems to indicate I should use MGCP which I would agree is a better choice but unfortunately the equipment I am trying to integrate only does H.248. Could anyone point me to something related to this. -- Chad Whitten Metro Network Solutions (601) 366-6630 Phone (601)

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

I'm attempting to connect to a Metaswitch, inbound only (at this time). The Metaswitch is the only "connection" (at this time). All I'm getting so far is a bunch of "OPTION" messages which my Asterisk box replies to but I don't get inbound calls. Here's my sip.conf. As you can see I've been trying a bunch of different options without success :(

I'm (still) trying to get my Asterisk box talking to a Metaswitch. All I'm getting is a "heartbeat" of OPTIONS messages coming from the Metaswitch which my Asterisk box replies to. The exchange looks like: <-- SIP read from 172.b.c.d:5060: OPTIONS sip:metaswitch@206.b.c.d:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP

What is the deal with "CSR's"? TWTelecom is telling me that I can't port a number to their service without a Customer Service Record. Apparently this is easy with Verizon, and not so easy with some other companies. Basically I'm at a brick wall with a couple of ports because TWTelecom is telling me I HAVE to get a CSR and certain other providers (Time Warner Cable for

Thanks Brian, and thanks again for the included definitions <grin> - that helped too. Your comments are really helping clear many questions. I suppose our intensions are to become an IXC. So if my local carrier is sporting old technology, they'll provide TDM services. So if I understood you correctly, the "in-band signaling" is typically SS7, and the alternative is typically

I am setting up VM off Metaswitch due to a problem with Metaswitch VM. I have a couple days to prove this works and I need a little assist please. I am using TRIXBOX 2.6.2.5 and have the Meta SIP trunk up. I have extensions built that can talk to each other. I took a trace on the TRIXBOX that shows when I dial my test phone on Metaswitch it goes to VM after a couple rings and the call goes to my

Hi folks, I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD 4.7 -release. Everything seems to work fine. I have a macro which answers, receives the fax to a tiff, and then runs a script (mailfax) to convert that to pdf and email it. It all works perfectly except for some errors I am seeing in the console. After it hangs up I get a dozen or so messages in the cli

Hello, I''m trying to port one of our existing appliances (running on 64-bit Red Hat Enterprise Linux 6.4) to a Xen HVM guest . We''re seeing some very odd behaviour that doesn''t manifest on other platforms. The guest experiences intermittent lockups of a few seconds- shell sessions become unresponsive, and various software healthchecking in our application triggers,

I've been racking my brain for the last two days to try to figure out what I could possibly be doing wrong in my configuration for a SIP trunk that's setup through my local ISPs Metaswitch. I've setup a very simple SIP Peer, which I've played around with a lot in the past two days but still comes back to the following basic setup: [provider-fireball] type=friend

Hi All, I'm stumped on a weird problem. I have an * server working fine for local SIP phones and IAX2 connections. We just provisioned a second Ethernet port to attach to a local SIP provider. PSTN calls incoming work fine: PSTN -> SIP Provider -> SIP -> * but outgoing calls are not. Call setup takes place and the caller can hear about 1-2 seconds of audio before the SIP provider

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Dear all what is softswitch what is difference between asterisk and softswitch ?? regards satish patel --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello averybody, I'm looking the softswitch in digium website, anyone test the softswitch? Best Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071202/2440f782/attachment.htm