[asterisk-users] TE121B server recommendation
Hello, If anyone is using a TE121B card and it works reliably (i.e. no HDLC Bad FCS or similar errors), could you pass along the make, model, and basic configuration of your Asterisk server? We tried upgrading our old Dell PowerEdge server to a SuperMicro system, but that didn't help. I would like a solid recommendation before I suggest another purchase. Thanks. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pre-routing manipulation of calls
This is concerning an Asterisk 1.4.18 server. We have approximately 70 DID numbers. Incoming calls are placed into the incoming context (by zapata.conf) and are routed based on the dialed number. I want to do some manipulation (CallerID name override) to all incoming calls before they are routed. I would prefer to avoid duplicating the necessary code in each DID extension stanza, even if it's just a call to a macro. 1. Can I set up a catch-all extension in incoming, do my processing, and then have the calls fall through to the existing extension stanzas? 2. Or, should I use a separate pre-incoming context to do the manipulation and then jump to the real incoming context containing the specific extension stanzas? 3. Or, is there another method that would be more elegant? Thanks. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pre-routing manipulation of calls
I use number 1 with a Gosub(get_name,s,1)
It jumps to a mysql lookup against the number and sets the name and
continues on.
Based on the ambiguity of the documentation with respect to extension
sorting order [0], I ended up going with the pre-incoming context
idea. It worked fine.
[pre-incoming]
exten =
_X.,1,Set(CALLERID(name)=${IF($[${DB(cidname/${CALLERID(num)})} = ]
?${CALLERID(name)}:${DB(cidname/${CALLERID(num)})})})
exten = _X.,n,Goto(incoming,${EXTEN},1)
By the way, I'm using the AstDB for CallerID overrides, which seems like
it would be more reliable than using an external database. Is there
some advantage (e.g. scalability) to using MySQL?
Thanks.
[0] http://tinyurl.com/3jn62a
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[asterisk-users] Tracking T1/PRI channel status - inbound vs. outbound
I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel x Or: Action: Status However, nothing in those results seems to reliably indicate whether the channel received an incoming call or was used to make an outgoing call. I can make educated guesses based on the value of Caller ID or Calling TON or the combination of Channel and Link, but none of these heuristics seem robust. I've even tried GetVar-ing various channel functions, to no avail. This seems like a case where a simple flag should be set somewhere, but I haven't found one. What's the most elegant way to do this? Thanks. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE121, echo issues, NMIs
(The only issue we've had with the TE121 is echo on voice calls, even with the hardware echo cancelling module and lots of zapata.conf tuning... did You EVER get the echo resolved ? How ? We managed to get it tuned to the point where user complaints are minimal, but there is definitely still a problem. We've also had Asterisk randomly die a few times. Nothing is written to the logs in these cases. Digium investigated the echo issue (over SSH) and claimed that my system is handling several non-maskable interrupts and that I should pull out the TE121 and watch /proc/interrupts. I can't really pull a production system down for that kind of invasive testing, so we're just living with it at the moment. If anyone knows of a particular server in which the TE121 is known to work *reliably*, I'd be much obliged. We thought that a SuperMicro server would be a decent choice, but apparently that's not the case. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it or another TDMoE bridge?
We used it in our installation and had some issues. We were passing fax and modem calls through via the second port as a TDM bridged call. For some reason, the timing was off even though we explicitly set the timing in the redfone.conf file. We replaced it with a Sangoma A102d and haven't been happier. We had many problems with the FoneBridge2-EC as well. We followed all of Redfone's hardware requirements (e.g. a good server, a dedicated NIC for TDMoE, a dedicated IRQ, good cabling, etc.) and even let them log into our Asterisk box to tweak things, but there were still many HDLC errors and glitches in the audio (irritating for voice and show-stopping for faxes). We replaced it with a Digium TE121 and the problems went away. Faxing has been clear and reliable ever since. (The only issue we've had with the TE121 is echo on voice calls, even with the hardware echo cancelling module and lots of zapata.conf tuning...) -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI - calling functions, CHANNEL STATUS broken?
I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). I ended up grabbing this info from the manager interface, within an AGI script. A little back-asswards, but it works. I still would like to know whether it's possible to call Asterisk functions from within AGI scripts... -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI - calling functions, CHANNEL STATUS broken?
This worked for me on * 1.6 where 1223 is the sip peer I wanted to get status from. use Asterisk::AGI; my $AGI = new Asterisk::AGI; my $peer = $AGI-get_variable(SIPPEER(1223,status)); It didn't work (for me) on 1.4.18. An empty string was returned, even though I gave a it a valid peer number. Oh, well. :-) Thanks. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI - calling functions [SOLVED]
Solved: use Asterisk::AGI; my $AGI = new Asterisk::AGI; my $peerst = $AGI-get_variable(SIPPEER(123|status)); my $peercc = $AGI-get_variable(SIPPEER(123|curcalls)); This works fine in 1.4.18. Thanks. -- Kevin DeGraaf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI - calling functions, CHANNEL STATUS broken?
Greetings,
I am writing an AGI script that needs to check on the idle/busy status
of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and
Snoms thrown in for fun).
Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI
scripts? Based on my Googling, I would guess in the negative. I have
tried various permutations of Set() and Eval() without success.
I have also tried to use the CHANNEL STATUS AGI command, but that
doesn't seem to work, as indicated by these results:
exten = 610,1,NoOp()
exten = 610,n,Set(CC_200=${SIPPEER(200:curcalls)})
exten = 610,n,Set(CC_221=${SIPPEER(221:curcalls)})
exten = 610,n,Set(CC_231=${SIPPEER(231:curcalls)})
exten = 610,n,AGI(test.agi)
$cc[0] = $AGI-get_variable('CC_200');
$cc[1] = $AGI-get_variable('CC_221');
$cc[2] = $AGI-get_variable('CC_231');
$AGI-verbose(Test using Set(): $cc[0] $cc[1] $cc[2]);
$AGI-verbose(Status of 200: . $AGI-channel_status('SIP/200'));
$AGI-verbose(Status of 221: . $AGI-channel_status('SIP/221'));
$AGI-verbose(Status of 231: . $AGI-channel_status('SIP/231'));
test.agi: Test using Set(): 0 1 0 ; Exactly as expected
test.agi: Status of 200: -1
test.agi: Status of 221: -1
test.agi: Status of 231: -1
Any advice?
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[asterisk-users] Polycom IP4000 - Device does not match ACL
I am trying to configure a Polycom IP4000 for use with Asterisk 1.4.5 on a flat local network. I followed the provisioning guides that I found on the Web, and I have the phone downloading bootrom.ld, sip.ld, and a bunch of configuration files. This all works properly. However, I receive the following error: NOTICE[27345]: chan_sip.c:14725 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for 'x.x.x.229' - Device does not match ACL I can place calls from the IP4000, but I cannot receive them: WARNING[27480]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Here are the relevant (IMHO) config sections. == sip.conf == [ip4000_1] [EMAIL PROTECTED] type=friend secret=password qualify=yes nat=no host=dynamic canreinvite=no == Polycom per-phone config on TFTP server == reg.1.displayName=207 reg.1.address=207 reg.1.label=207 reg.1.type=private reg.1.lcs= reg.1.thirdPartyName= reg.1.auth.userId=ip4000_1 reg.1.auth.password=password == Polycom company-wide config on TFTP server == server voIpProt.server.1.address=x.x.x.55/ SIP outboundProxy voIpProt.SIP.outboundProxy.address=x.x.x.55/ /SIP I've tried using x.x.x.55 as both the proxy value only, the server value only, and (in the given example) both. I also added the following to sip.conf, to no avail: deny=0.0.0.0/0.0.0.0 permit=x.x.x.0/255.255.255.0 Any ideas about what I've missed would be appreciated. Thanks. -- Kevin DeGraaf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] foneBRIDGE2 vs. foneBRIDGE2-EC
Hello, I'm trying to decide between the foneBRIDGE2 ($1135) and foneBRIDGE2-EC ($1610). Has anyone here directly compared the two? Would we really suffer without the onboard echo cancellation? The manufacturer's site doesn't really give much helpful information about choosing one over the other. Thanks. -- Kevin DeGraaf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple lines on Linksys/Sipura phones
I'm going to be deploying around 30 IP phones with Asterisk in the near future. I've tentatively settled on the Linksys/Sipura SPA9xx family. I am unclear on the notion of lines in the context of SIP phones like these. The SPA942 model has a 2-line-to-4-line upgrade available, but I don't know why I'd need to purchase it. I have tested a SPA942 with Asterisk, and even without the upgrade, I can easily send/receive/hold four separate calls at a time, using the four line keys. What am I missing? Thanks. -- Kevin DeGraaf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detect phone pickup, caller ID AGI
Hello all, I have Asterisk running as a simple voicemail server. I don't have any FXS ports, IAX/SIP clients, anything like that, just an X100P attached to my phone line in parallel with a number of normal phones. I have a Wait(18) command in my dialplan which works fine; Asterisk picks up after four rings and proceeds to record voice mail. However, if I pick up one of the regular phones, Asterisk still proceeds. Can my X100P/Asterisk solution detect a phone pickup and abort, just like a normal answering machine would do? Also, is it possible to set up the dialplan such that an AGI is always called, regardless of whether another phone is ever picked up, as soon as Caller ID data is available (via a method more elegant than a Wait statement)? (I'm interested in having MythTV display the Caller ID information upon an incoming call, as soon as Asterisk has decoded it.) -- Kevin DeGraaf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
