[asterisk-users] Split asterisk in two ?? One TDM and One IP only??

I have built an asterisk server with a TE412P card on a Dell 2950. It does incoming calls via DID over PRI, our IVR, SIP/IAX extensions, Fax/Analog extensions via an old PBX via PRI, voicemail, etc. My issue now is that I find it difficult to test/upgrade to new versions. This is what I am

[asterisk-users] FYI about my Mona Vie business venture

Asterisk work does not pay all of my bills, so I have joined up with a company that has a very good payment plan. I have recently become a Mona Vie Independent Distributor. I am not going to go into a sales pitch. This is just an FYI to this opportunity. The company has grown into a Billion

[asterisk-users] FYI about my Mona Vie business venture - apology and rethink

I feel the need to apologize for my previous email to the list. I was thinking that I was sharing something that I am currently exited with, with my associates. I now realize that I was sooo... off-topic, it's ridiculous. Sorry for the improper post to the list. Feel free to keep any further

[asterisk-users] asterisk linkedin group

asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Please visit us on the web at

[asterisk-users] meetme list (unmonitored)?

What does unmonitored mean in the below reference? Ref: CLI meetme list User #: 01 5665 zzz Channel: SIP/5665-9f8038a0 (unmonitored) User #: 02 5664 no nameChannel: SIP/5664-0096b660 (unmonitored) 2 users in that conference. Also, is

[asterisk-users] Announce option for meetme - is it used?

Announce option for meetme - is it used? It makes a caller record their name, but I do not see where this name recording is ever used.   Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Fax. 248-836-5101 www.hirotecamerica.com Board member of

[asterisk-users] iax vs. sip?

I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am

[asterisk-users] All circuits are busy now???

"All circuits are busy now" makes perfect sense in my PRI trunk is full. How do I stop asterisk from playing this recording when it is a wrong/bad number? I gat a call today that a user was trying "all day" to call a number in Mexico and kept getting the above recording. I said, try

[asterisk-users] Setting QOS settings in asterisk and/or CentOS?

How would I go about setting the TOS bit to "RTP IP TOS Byte: 18 (hex)" for SIP and IAX traffic at the asterisk server? Also, Do you have a quick reference on how to configure a Cisco switch to prioritize SIP traffic? I check in various Cisco docs, and there are so many references, and

[asterisk-users] hint status from dialplan?

Can I get the hint status from the dialplan? I am intending to add lit buttons for the parking slots. currently I am using 1.2.11 with 1 parking button and several pickup buttons (speed dials to the parking slots) since 1.4 allows park() to specify a parking slot, I figure that I can remove

[asterisk-users] Odd echo issue with speaker phone

I assume that this is from the echo canceller, but I am not sure. A call is started via SIP speakerphone. When the handset is picked up, there is a slight echo of your own voice after you speak.(duh, is there any other kind of echo) If the call is made without the speaker phone, there is

[asterisk-users] Odd echo issue with speaker phone

I assume that this is from the echo canceller, but I am not sure. A call is started via SIP speakerphone. When the handset is picked up, there is a slight echo of your own voice after you speak.(duh, is there any other kind of echo) If the call is made without the speaker phone, there

[asterisk-users] Inhouse SIP to ZAP has echo sometimes.

Sometimes we get echo heard on SIP phone when dialing out. Zap channel is on aTE411 Card. It is using a PRI to XO. As far as I know echo is created on the far side. Could the Zaptel card be the far side as far as the SIP phone is concerned? Calls from our soon to retire legacy PBX do

[asterisk-users] some transfers dropped.

We are having an issue with transferred calls being dropped. Looking at the asterisk 1.2.10 logs, it appears that when it is dropped, the SIP unit send a CANCEL message to the server. On successful transfers this is not seen. The errors logged in the SIP Unit error log, I believe are from

[Asterisk-Users] revisit to legacy PBX and CID over PRI

My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will

[Asterisk-Users] Cell gateway for T-Mobile US??

Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product

[Asterisk-Users] Re: CallerID name inbound from PRI

XO fixed my caller ID name. I am using FreePBX and I can include a wait to my custom extensions. Is there a way to add a wait to the whole PRI? I assume that if I set immediate to yes, I can then have a s extension do the wait, but how would it get from the s to the DID extension? (also, I would

[Asterisk-Users] calleridname.agi patch to only overwrite name if it is missing

I edited the calleridname.agi patch to only overwrite the name if it is missing. The asteridex option still overwrites the name since it is our master list for known numbers. -- Steven calleridname.agi.patch: --- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13

[Asterisk-Users] Backup Question?

This may be slightly off topic. I am using FreePBX, and using it's backup feature. Here is the question part: I would like to copy my backup off the asterisk server. From your experiences, which approach seems more resilient to failure: Push the backup from asterisk to another server

[Asterisk-Users] me, voip.trxtel.com and early media

I was testing using trxtel for outbound toll free because I have an issue on my PRI where it will not handle early media. (IVRs that play as a ringback tone) There was a bug that was supposed to fix this Q4 of 2005, but I never saw any relief for it. voip.trxtel.com has the same issue, so at

[Asterisk-Users] SE Michigan asterisk users group

I am thinking of getting an asterisk user group together for either SE Michigan or just Metro-Detroit. How much interest in asterisk in Michigan is there on this list? I am already on the board of glimasoutheast, with is a group for technology professionals. (very broad range) It is a spin-off

[Asterisk-Users] metermaid patch

Olle, Will the metermaid patch help this issue?: http://bugs.digium.com/view.php?id=7435 I believe that the fix is in res_features.c , but I do not want to pursue it if it is already there. Also, thanks for your hard work on that patch. Our receptionist will really like the PARKINGEXTEN

[asterisk-users] IVR with LDAP query for phone number and mobile number??

We still have a lot of users on a legacy PBX, so the Directory app is not sufficient. We also have users with mobile phones. Has anyone made an LDAP lookup that will pull this info from MS Active Directory? My thinking is to add this function to my main IVR. As long as my AD is accurate, it

[asterisk-users] nat and qualify questions

Are there any problems with always having nat=yes and qualify=yes? We just opened up our server to be accessible to SIP from the internet. (used to require VPN) I had to set the SIP setting for my test softphone to nat=yes and qualify=yes. This makes sense. Some of these phone will

[asterisk-users] Iaxy and SendDTMF??

I have an Iaxy that I am using to access our overhead paging system. It is ext 5480 and required a 1 (office), 2 (Shop), or 3 (all) DTMF tone after it answers. If I dial 5480, I hear a tone to let me know that it is ready for the digit. I made an extension 5481 that using a macro and sendDTMF to

[asterisk-users] dial D option with w for wait?

From WIKI: D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel. (You can also use 'w' to produce .5 second pauses.) When I use the D option to send a call to my paging system and pick a zone, the Tone is too early. I have

[asterisk-users] VM notification to pager and phone

I looked for a reference to do this for some time to replace the callout feature in my old AVT voicemail. I never found one, so I decided to dig in. Here is my first run. It is in production, so unless I find a problem, I am done. Script set to run every 5 min. via cron. This sets a lock

[asterisk-users] metermaid and 1.2.13?

It is unclear to me if the metermaid patch should be in 1.2.13 or not. Please advise. Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org

[asterisk-users] Vonage SIP access via asterisk?

Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some

[asterisk-users] asterisk linkedin group

asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Board member of Connectech Greater Detroit www.connectech.org

[asterisk-users] best way for night ringer??

Asterisk 1.2.13 I am trying to figure out the best way for a night bell at work. Note: I have no spare buttons available on the phones. But I do have two lines and two park positions as buttons. Option 1 (easiest and the one I just implemented) When asterisk is in night mode,