TDM410 and DSL

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Hi all,
I have a system installation in Guam with two trunks. One has a DSL service
riding on it with the usual filter. That channel however keeps throwing
alarms. I bypassed the filter and it stopped throwing alarms, but of course
the high frequencies annoy the users. I swapped the filters and the alarms
came back. Any suggestions? Could I have a bad DSL modem? Cassius

Asterisk Users 4.6 years ago 0 Answers

Too Few Fax Detections

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OK, after my last message about fax detection, I feel a bit better informed and able to press forward. I started looking into this because I was getting lots of false positive fax detection errors in the logs with faxdetect=both set in chan_dahdi.conf. Anyhow, I do not currently use fax detection, and we have a dedicated Fax DID on our PRI, so setting faxdetect=no works fine. Having said that, I would like to sort it out as I may want to use fax detection in the future. Unfortunately, I seem to be having odd results. I set faxdetect=incoming last night…

Asterisk Users 4.6 years ago 2 Answers

Weird phone behavior after recent CentOS 5 update

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For some reason our Asterisk box is doing something really unusual following applying a routine update to CentOS 5 on Monday. We have Asterisk 1.4.2 and its been working great for years. But now when the phone system receives an incoming SIP call, its not providing any audible dial sound to any caller. It is recognizing the incoming call, and after no answer for about 5 rings or so, it goes to voice mail. But there is no audible 'ring' to the caller. Just nothing - blank, empty silence. Of course any automated answering system (ie. business phone menu, etc.)…

Asterisk Users 4.6 years ago 2 Answers

isr2=XX isr3=Y

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I have just built a new system using an HP DL360G7 with a TE420 T1 card,
and this is the first system using a generation 7 server. I'm not sure
whether that is an issue or not. I am using Asterisk 1.2, and Zaptel 1.4.12.1 with patches for GEN5 of
the TE420 card. I have successfully used this combination on several
systems based on the DL360G6 and TE420(gen5), which have been in
production for many months now. However, on the new server, while doing loopback testing, I have found
that several minutes after starting…

Asterisk Users 4.6 years ago 0 Answers

Blind Transfer not working - 1.4.38

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Hi We've been running asterisk 1.4.17 (deb package) in a production
environment for some while now and are finally taken the plunge to
update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
Architecture I have upgraded the asterisk version in one of our test environments and
blind transferring seems to have suddenly stopped working. It was
working fine under 1.4.17 So, call comes in to extension 501 who does a blind transfer to
extension 504 at which point the call gets completely cut off. I ran a SIP trace of this…

Asterisk Users 4.6 years ago 3 Answers

Post Dial Delay + Playtones

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Can anyone give me some pointers on the following, in our setup (ast
1.6.3) we use international carriers to terminate calls for a
callingcard system, we have an issue where there can be a very long
delay after dialing but before the far end begins to ring. I would like to play a tone every second during this period (before
ringing) but then cut off the tone once the far end either sends ringing
or progress in band (183). Iv tried using Progress() and Playtones() before the dial but Playtones
cuts off as soon…

Asterisk Users 4.6 years ago 0 Answers

id to SIP-invite

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Hello list, is it possible to add the field Privacy: id to a SIP INVITE message ?
INVITE sip:32444666600@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP1 .2.3.4:5060
From: "R3211133333" ;tag=2096790244
To:
Call-ID: href="mailto:3b040826e909d311880a0090330605a0@192.168.12.40">3b040826e909d311880a0090330605a0@192.168.12.40

CSeq: 34677 INVITE
Contact:
Allow:
REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE
Content-Length: 203
Content-Type: application/sdp
Max-Forwards: 69
Supported: replaces,answermode,100rel
User-agent: (innovaphone IP800/6.00 sr2-hotfix16 [09-60901.35/424/110])
*Privacy: id* How can I do this in the Asterisk dialplan ?? SIPAddHeader ??
Kind regards,
Jonas.

Asterisk Users 4.6 years ago 1 Answer