Re: [asterisk-users] Polycom UC 4.x Unreachable
Solved it! Turns out UCS Polycoms are quite picky about blank callerids, to the extant they ignore those packets completely. My global "callerid=" in sip.conf was intentionally blank. In ten years, in never caused a problem. By setting to 0, the Polycoms that didn't respond to SIP OPTIONS (nor the NOTIFY for waiting messages) now work fine. If anyone is curious, the problem is easily reproduced in the dialplan by setting the callerid there to blank, then the UCS polycom will ignore that INVITE as well. Set the callerid to anything else and it'll ring. On 23 August 2017 at 19:29, John Covici <cov...@ccs.covici.com> wrote: > I always set it to no, but set the registration time to 60 seconds, > and that has always worked for me. > > On Wed, 23 Aug 2017 17:23:38 -0400, > Gary Reuter wrote: >> >> Hello, >> We've had dozens of Polycom 3.x firmware phones deployed and working >> great for years. >> Now I've finally been charged with the long-overdue task of figuring >> out why newer Polycom devices with 4.x firmware register fine but do >> not respond to SIP OPTIONS request and therefore always become >> UNREACHABLE if the sip qualify setting is set to yes. >> >> To my dismay, searches for solutions from others who have encountered >> this problem have given zero results. >> >> >> Thanks! >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > cov...@ccs.covici.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom UC 4.x Unreachable
Hello, We've had dozens of Polycom 3.x firmware phones deployed and working great for years. Now I've finally been charged with the long-overdue task of figuring out why newer Polycom devices with 4.x firmware register fine but do not respond to SIP OPTIONS request and therefore always become UNREACHABLE if the sip qualify setting is set to yes. To my dismay, searches for solutions from others who have encountered this problem have given zero results. Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off Topic
Please forgive this off-topic post... I've been on this list since 2005 (over 45k messages in my archive) and this is obviously really not something I normally do. If you have a minute and are feeling generous, please visit http://bailout.chipin.com/ and consider helping me out. Sorry if I've offended or wasted your time, but believe me that you don't feel as bad as I do. ___ -- 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] ECHO Tutorial
On 6/19/06, Daniel Salama [EMAIL PROTECTED] wrote: Is there anyone that could explain to me the phenomenon of Echo or at least point me where I can learn more? This paper by Cisco is a great start: Echo Analysis for Voice over IP http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml (it's the first result I get when I google for echo in voip) Why is this affecting the VoIP world so much and not the regular PSTN analog world? That's a misconception -- echo is always there, just not perceivable. (I actually think that if absoluteley all echo were removed, conversations would sound strange, similar to being in an anechoic chamber.) What does the PSTN industry have that they can handle such high volume of calls and there is no echo problem? Expensive equipment with built-in echo-cancellers? Extensive planning and testing before deployment?Decades of experience dealing with such problems? ___ --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
Re: [Asterisk-Users] SIP 486 Busy Here
On 6/9/06, Jason Lixfeld [EMAIL PROTECTED] wrote: Kinda confused by this... I have a Cisco 7960 configured with a couple SIP extensions configured on the phone. Just trying to dial one extension from the other on the same phone, but when I do, I get: Could the phone be returning 'busy' because you are on a call in dial-state (as opposed to an established call)? ___ --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
Re: [Asterisk-Users] Connect 2 Asterisk Servers via PRI
On 5/31/06, Bruce Reeves [EMAIL PROTECTED] wrote: Both servers are runing Asterisk 1.2.7 and the T-1 is a cross over cable less then 10 ft long. I figured the pri_cpe and pri_net were right, but there must be other problems. Any help is appreciated. Is there a specific reason you are connecting the servers in this way? Have you considered using IAX, SIP, or even TDMoE instead? Two main advantages: 1) Lower cost per port -- adding a dedicated ethernet card is way cheaper than using a T-1/PRI port from anything that's supported by Asterisk. 2) More features -- the current functionality of libpri/asterisk does not support all PRI functions across all switchtypes completely. For example, 2BCT (call transfer) was only functional for 5ESS switchtype last I checked. ___ --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
Re: [Asterisk-Users] MySQL replication for voicemail
On 5/9/06, Damon Estep [EMAIL PROTECTED] wrote: How many users are you supporting on a single 8 box using obdc/mysql voicemail storage? Performance issues? If you are concerned about your system's performance, then do NOT use ODBC message storage! Reading through app_voicemail last weekend, I realized that the ODBC storage code writes/erases the recordings to disk temporarily everytime it is required -- the recording fetched from the database is not streamed from memory. Also note, the current app_directory has absolutely no code for ODBC storage, so user's name recordings do not get used -- names are always spelled out. ___ --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
Re: [Asterisk-Users] MySQL replication for voicemail
On 5/9/06, Damon Estep [EMAIL PROTECTED] wrote: I had similar concerns, but wanted to get a feel for what others are experiencing. The extra overhead shouldn't really be a problem on a decent machine, especially with a ramdisk-bask voicemail spool. The goal is to be able to extend our web portal to be able to view/listen/delete VM without having to put the load on the * box. I have mysql database (realtime and voicemail storage), web, and ftp (provisioning) on a seperate machine from asterisk. Rewrote the vmail.cgi in PHP and everything works fine. Already using realtime mysql for VM config. I had to make asterisk use ODBC for everything when I converted to ODBC message storage. res_odbc and res_mysql wouldn't co-exist. There's a php script on the wiki to load up your voicemessage table with any messages currently on disk... ___ --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
Re: [Asterisk-Users] MySQL replication for voicemail
I have voicemessage ODBC storage working and MySQL replicating, but I didn't setup the replication after-the-fact, therefore didn't need the 'LOAD DATA FROM MASTER'. You may want to try a different method (more manual) of synching your machines. Does your voicemessages table contain specially large messages? The errors you get don't suggest this, but the only change I had to make was to increase the maximum packet size -- it was 1M by default and I changed it to 16M. On 5/8/06, Noah Miller [EMAIL PROTECTED] wrote: Does anybody have any ideas on what's causing this error? Why would MySQL not have enough memory? What does it mean, points outside data file? Is anybody else doing this successfully, or am I a lone freak? ___ --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] ODBC Voicemail storage and app_directory
I've checked the wiki, searched the mailing list, Mantis (bug tracker), and looked through the source code, but found not mention of this issue: If using ODBC Storage for voicemail messages, name-playback by app_directory breaks. app_directory has no ODBC Storage code and therefore only looks for the greet.wav file (recording of user's name) in the filesystem, whereas app_voicemail saves the recording in the database. (1.2.6 stable) I'm hopnig I missed something somewhere and someone will point me to a patch or the bug number in Mantis! ___ --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
Re: [Asterisk-Users] OT: MWI on Treo 600/650
On 4/13/06, David Cook [EMAIL PROTECTED] wrote: My cell vm goes to asterisk, not the carrier. Apparently MWI is turned on/off with specially formatted SMS messages. Anyone know how to do this on a Treo 600? Having the phone light from Asterisk would be HUGE ... not to mention extremely cool. I've had this working using Clickatell.com as a third-party provider, and by using Gammu (gammu.org) with a Nokia 6190 (using serial connection). MWI is set by setting the UDH of the SMS to '04010200xx' where the last two digits indicate the number of new messages -- when 00, the MWI is turned off, other values should display on the phone. BTW, not all cel-carriers allow you to change the message-center number for voicemail -- they've disabled the menu option in the phones to prevent you from changing it. ___ --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
Re: [Asterisk-Users] Polycom NTP issue
I have several Polycoms (301, 501, 601) all working fine -- some with sip ver 1.5.2, others recently upgrade to 1.6.5. The only problem I've had was incorrect time for the first few minutes when the phones boot-up, but that's been fixed by Polycom in newer versions. The SNTP section of sip.cfg in 1.6.5 has dhcp-override settings which weren't there in 1.5.2, but my guess is that your 'gmtOffset' is not set right... Below is the section from my sip.cfg that sets GMT-5 (Eastern)... SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=clock.redhat.com tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset=-18000 tcpIpApp.sntp.gmtOffset.overrideDHCP=0 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1/ On 4/27/06, Kerry Garrison [EMAIL PROTECTED] wrote: Polycom 501 Firmware: 1.6.2.0041 Bootrom: 3.1.0.0269 ___ --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
Re: [Asterisk-Users] PRIs from two different telco
I have 3 PRIs from different providers, and the fourth port is to a legacy Meridian system... Is it always the same PRI that does not work regardless of which port is used on your Digium card? I don't know if it's significant, but the order of my zaptel.conf is different than yours: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,2,0,esf,b8zs bchan=49-71 dchan=72 span=4,3,0,esf,b8zs bchan=73-95 dchan=96 (Note, I do not have the echo-cancelling card.) On 4/27/06, Wai Wu [EMAIL PROTECTED] wrote: I just tried it. Same problem, one of the two spans is not working. If I load wct4xxp with vpmsupport=0, then both spans working. BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 loadzone=us defaultzone=us ___ --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
Re: [Asterisk-Users] channels change names
Another option would be to set/pass a variable and use that instead of the current channel variable value. Keeping the Local chan up just to maintain a constant variable value may be alot of overkill (longterm) compared to rewriting your code to set and use your own variable (shortterm). On 4/20/06, Peter Fern [EMAIL PROTECTED] wrote: Probably because the Local proxy channel drops out once the two sides have been bridged. If you want the Local chan to stay up, use the /n parameter and the local channel won't perform the native transfer. This does have it's own problems, but should do what you want. eg: Channel: Local/[EMAIL PROTECTED]/n Jon-o Addleman wrote: I'm writing a php script to dial numbers and connect them to a conference. This is fairly straightforward: Action: originate Channel: Local/[EMAIL PROTECTED] Context: default Exten: $extension Priority: 1 This is pretty straightforward. However, the script then loads the list of members in the conference (using the meetme list ... command). For local extensions this works fine - the list of members shows the right channels, etc. The problem I'm having is that if the extension is external, the conference list shows a Local/$extension channel at the start, and then once the call is completed, it changes the channel to whatever was dialed. I'm probably not explaining it properly, but what I'd like to have happen is that I get one consistent channel name from the start of the connection - it doesn't matter what it is, as long as it doesn't change. As things stand, the conference list isn't accurate, unless I wait about 5 seconds after adding someone before updating the list. Thanks for any suggestions you might have here! ___ --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 ___ --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
Re: [Asterisk-Users] SIP Subscriptions
Doesn't realtime allow you to manipulate users without ever having to issue a reload command?I'd expect the command 'reload' to flush any existing information from memory and use with freshly loaded information anyway, and that goes for any application, not just asterisk. On 12/20/05, Douglas Garstang [EMAIL PROTECTED] wrote: So I see that if you issue a 'reload' command in Asterisk 1.2.1, you lose all your SIP subscriptions. Nice. Basically that means the use of hints and subscriptions in a production environment is a completely impossible. Awesome. Considering traditional phone users have come to expect this functionality, it leaves a lot to be desired as far as Asterisk is concerned. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
Re: [Asterisk-Users] Attack dialing
On 12/11/05, Michael Welter [EMAIL PROTECTED] wrote:
I think attack dialing means to dial all 10,000 number in an exchange,looking for modems and fax machines.BTW, Colorado Springs, Coloradohas made it illegal to dial a number without intending to have a
conversation sighProbably something to do with NORAD or Space Command.
It's actually called 'war-dialing'. There were loads
of programs to do it using a modem back in the day -- they'd even
randomize and track the dialed numbers.
But Eric just seems to want an auto-redial-on-busy/congested -- which should be pretty simple...
Assuming you have Zap PRI as group 1 and the specific number you want to redial is 5551212:
exten = 5551212,1,Dial(Zap/g1/${EXTEN}
exten = 5551212,2,GotoIf($[x${DiALSTATUS} = xANSWER]?10)
exten = 5551212,3,Wait(3)
exten = 5551212,4,Goto(1)
exten = 5551212,10,NoOp(call was answered so do something with it)
If anyone knows a way to detect if a remote number becomes 'un-busy'
without actually dialing the number, it would make for an even more
elegant solution.
___
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Re: [Asterisk-Users] PRI to SIP Problem
Yes, many people have had this problem. Check the mailing list archives... I think the newest code has the fix. Workaround for older versions is to Answer before Dial, but you may still need the 'r' option to Dial as ringing may stop for the caller after about 10 seconds.On 10/27/05, OTR Comm [EMAIL PROTECTED] wrote: Hello all,I have a problem calling into asterisk on a PRI going out to a SIP phone(PRI - SIP).The calling party does not hear ringing and after about fiveseconds gets an *All circuits are busy* recording.However, the called SIP phone does ring, and if the called party answers the phone within a fewseconds, the call stays in service.CLI messages:...-- Accepting call from '9288532045' to '6023432727' on channel 0/23, span 1-- Executing Dial(Zap/23-1, SIP/102|20|rt) in new stack-- Called 102!! Don't know how to add an IE High-layer Compatibility (125)!! Unable to add IE 'High-layer Compatibility' -- SIP/102-935d is ringing-- Channel 0/23, span 1 got hangup request== Spawn extension (incoming, 6023432727, 1) exited non-zero on 'Zap/23-1'-- Hungup 'Zap/23-1'...NOTE: There is no problem calling from SIP phone out (SIP - PRI). Any body ever have this problem?Thanks,Murrah___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming CallerID Name display
On 10/26/05, Andy Vega [EMAIL PROTECTED] wrote: I'm running a T1 between Asterisk and a Nortel Meridian Option 61c. When I call from Asterisk to the Nortel system, it displays both the name and number of my SIP phone. When I call from the Nortel system to Asterisk, I only get CallerID number. Set-to-set calls on the Nortel system display both name and number. Is it possible to get CallerID name from a Nortel switch? I'm running Lucent 5ESS emulation with ND2 signaling. Andy. I have a Norstar MICS and was unable to find a way to make it send out the name as well as the number. Workaround: Have each handset have it's own OLI, and then use Asterisk's LookupupCIDName, or use a macro to do the lookups in the realtime voicemail db and use SetCIDName. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 601 XHTML microbrowser
On 10/26/05, Chris HARIGA [EMAIL PROTECTED] wrote: I have a show parked calls php script for my Polycom IP600 phones. If U are interested let know and I can email it. Even if Sean doesn't want it, I do! All examples can be helpful. :-) Why not put up a page on the wiki linked from the polycom page(s)... If formatting is problematic, just note it on the page and I (and others) can help make look nicer for the wiki. -Gary ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] strange behaviour of asterisk sip.conf type=user vs type=peer
My guess is the problem is related to the reverse lookup of 147.135.20.128. sip.broadvoice.com = 147.135.20.128 147.135.20.128 = non-existant If you change your host=sip.broadvoice.com to host=147.135.20.128 it'll probably work until they change the IP of their server Someone else who actually uses broadvoice may have a better solution. On 10/25/05, Vikas [EMAIL PROTECTED] wrote: Hi,In sip.conf iftype=peerthen incoming calls from broadvoice are sent to context=incoming inthe extension.conf file.But in sip.conf iftype=userthen we get the following messages in the packets being exchanged with broadvoice:1. Found no matching peer or user for '147.135.20.128:5060'2. Looking for 4155131083 in default3. Reliably Transmitting (no NAT):4. SIP/2.0 404 Not Found The sip.conf host section for broadvoice looks like:[sip.broadvoice.com]callerid=xxcontext=incomingfromdomain=sip.broadvoice.com host=sip.broadvoice.cominsecure=verysecret=type=peeruser=xusername=xWhy is this happening,vikas___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] clarification please: accountcode, pri channel groups, and CDR
Hi, I have a question regarding the behaviour of the 'accountcode' setting in zapata.conf for PRI channel groups: I'm assuming the accountcode can be different for each channel-group, and applies just like the context (i.e. the accountcode setting for a channel group will have no effect when this group is used for outbound calls). I have TE405P with all four spans in use, three to telcos, one to a legacy PBX. For incoming DIDs on the telco spans, I need to use SetAccount if I want to differentiate. Will all calls initiated by the legacy PBX span use the accountcode for this channel-group (same as it does context)? Does the 'group' for the channels apply only to outbound (ie Dial/gX) and the context and accountcode apply for calls inititated by the channels in the group? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PROBLEM WITH A PRI INCOMING CALLS
What kind of phone do you have on the asterisk side? Have you tried just having the call from the Meridian go directly to voicemail? If you hear the announcement and can leave a message, the problem isn't with your ISDN connect, but on the other side. If it is, turn on isdn debug (pri debug span...) and place one working call, then place one non-working call (the other way) and compare the output.On 10/24/05, Alvaro Parres [EMAIL PROTECTED] wrote: Hi list, i have the next situation I've a asterisk connect with a Nortel Meridian Op 11, via a PRI CARD with 5ess switch [Asterisk] -- PRI - NET --- PRI- CPE --[Nortell] I can call from Asterisk to Nortell with no problem, but when Nortell place a call to me, i have the channel bridge but no audio can hear y any way.. Any idea ??? thanks. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Red Alarms, No D-Channels, and Crazy People
Why not attach /etc/zaptel.conf and /etc/asterisk/zapata.conf? On my TE405 (PRI, not plain T1), a red alarm detected by software (kernel, zttool, or asterisk), coincides with the red led. First thing I'd actually check is the wiring: if you jiggle the cable and the led changes, you've got a serious problem, but since you've only been there 3 weeks, you can blame on the previous guy! ;-) On 10/24/05, David Tillman [EMAIL PROTECTED] wrote: well post your setup from the provider and maybe someone can help.I'm not sure I follow you. More detail: We only do VOIP in-house. The asteriskbox connects to a T1 provided by SBC by way of a Digium Wildcard TE110P. Zaptel.conf indicates that the last admin believed the T1 to be configured forb8zs and esf.In the 3 weeks that I have been here, we have had a few dropped callsdaily. However,today has been a disaster. Thanks,-dave___ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI stopped accepting calls
Hi, I have an asterisk box with a TE410P (quad pri) which has 3 spans in use, 1 and 3 to two different telcos, span 2 to a legacy Norstar MICS. Everything has been working fine for months, but early this morning, the 1st span stopped accepting incoming calls, but outgoing calls on this span still worked. Nothing in the logs indicates a change overnight, span reset, or anything else obvious, excep what I've noted below: - every hour, the b-channels were restarted without any indication of a problem - chan_sip stopped logging the 'retransmission' messages just before 6:13 b-channel restart - first failed incoming call at 7:40:50, 7:41:01 has log entry span 1 got hangup request - 8:13 b-channel restart logged extra Got restart ack on channel 0/20 span 1 with owner - 8:43 first successful outbound call on span1 -- no evidence or reports of any problems with outbound calls on span1 - 8:55 I started diagnosing problem - 9:03 successful inbound call on span 3 (different telco, different switch-type) Until 9:05 when I forced a restart, no inbound calls succeeded on span 1 -- all of them logged a 'hangup request' 11 seconds after the initial call setup. After searching the list archives, the only mistake in my config I've found is having both span 1 and 3 set as primary timing source (instead of having one of the set as secondary). /proc/zaptel/? shows that span3 is the timing source (since the restart, don't know about before). Could this be the source of the problem? Why wasn't I affected before? BTW, I've been restarting asterisk every 3 or 4 days as a preventative measure -- the current uptime for the process was less than 36 hours. Thanks for any light that can be shed -Gary ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI calls to Automated Attendants Dropped
Sounds similar to a problem I've seen with a slightly different setup Calls to certain AA/PBXs were not passing progress information beyond 10 seconds into the call. Can you check your logs for the exact amount of time after the setup that the call gets dropped? I'm guessing you'll see 10 or 11 seconds for most calls. The work-around is to make the asterisk box doing the relaying Answer() the calls before doing the outbound Dial(). On 10/13/05, Dave Wise [EMAIL PROTECTED] wrote: I have 2 * boxes.1 has 2 PRI's from the Telco, and a PRI to the 2nd *The other has ZAP channels to Channelbanks for endusers.If someone on the second box calls a Toll Free number (it probablydoesn't matter that it is toll free) that is auto answered by an auto attendant (QVC, a Bank, the Airlines, Credit Card Companies) thenthe call gets dropped with in a couple of seconds of placing the call(the auto attendant barely gets started).Has anyone ever heard of this?I heard of people not hearing the auto attendant on some systems(not asterisk) because the channel isn't cleared or accepted (some sortof signaling related to these auto attendants).(maybe the samesignaling that shuts off the audio on some PBX's is hanging up the *???).Any ideas/solutions?___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MWI integration between Asterisk and Meridian
I've been wanting to do exactly the same thing, but I believe it's beyond my coding skills. I think we need a function similar to the SirrixMWI. Some initial code for MWI exists in libpri, but nothing in the rest of asterisk calls those functions yet. On 10/12/05, kritikus Araklidas [EMAIL PROTECTED] wrote: I try to integrate my old PBX Meridian and Asterisk througth a PRI T1 (I'mgonna use only the asterisk voicemail system) but i don't know how tointegrate the MWI protocol between Asterisk Voicemail and my Nortel meridian.Anyone know what i have to do for that.?Any idea is appreciated.Regards.Cristian._Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remotely dialing calls from a polycom phone
This looks like the info you want: http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config BTW, is your touchpad app publicly available? On 9/29/05, Eric Lawman [EMAIL PROTECTED] wrote: I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is fine, except it can be confusing for the receptionist. Why do I have to answer the phone so I can page someone? What I'm looking for is a little more direct integration between the touchpad and the polycom. So, the question is, does anyone know of a way to initiate a call on a polycom via the remote access port. I was thinking of something similar to the way you can reboot one using a NOTIFY message containing check-sync. I've tried monitoring the sip messages between the phone and asterisk, but so far have only succeeded in making a phone call itself. Any suggestions would be appreciated. Eric. ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pridialplan per call or per channel group?
Hi, Is it possible to set the pridialplan (or prilocaldialplan) on a call-by-call basis, or for a particular PRI channel-group? >From my experimentation so far, it appears to me that the pridialplan and prilocaldialplan are global in zapata.conf, unlike other options such as switchtype, signalling, and context. I currently have two PRIs from different providers (and different switchtypes), and a third PRI port going to a Norstar MICS 4.x. I'm trying to get better dialplan integration with the Norstar by using some of the features described at the beginning of the installer's guide (ie, using 'private' or 'tie' line to link two remote MICS with public-network PRIs at each end). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] passing variables to h extension
I have something similar to do SMS voicemail
notifications... I do not use any underscores when I set
the variable and it works fine in the 'h' extension.On 9/13/05, Simone Cittadini [EMAIL PROTECTED]
wrote:Is there a way to pass variables/arguments to the h extension ?
for example :[default]exten = _1098933X.,1,NoOp(CARRIER TWT-TIM, EXTEN: ${EXTEN}},SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN})exten = _1098933X.,2,SetVar(_PROVA=bla)
[lot of stuff, agi, goto, tricks and magic that happens]exten = _1098933X.,10,Dial(${CHAN_DEST},,L(360:3599900)) - don'tmind L, a quick hack for dtmf not working with sipexten = _1098933X.,11,Hangup
exten = _1098933X.,12,Playback(no-credit)exten = _1098933X.,13,Hangupexten = h,1,NoOp(${PROVA})When the calls hangup, no bla is printed on screen, I think it's fine,
since the variable is automatically trashed when the channel ishungup., sigh ...But I need to pass some variables from the calling extension to an AGI,like :exten = h,1,DeadAGI(
update-credit.py|${CALLER}|${CALLED}|${CARRIER})in order to decrement the amount of credit for each customer after everycall.I've seen that in others prepaid systems built over asterisk theupdating of available credit is done in a cron job, have I to take it as
a sign that real-time billing is impossible ? Hope I haven't to ___--Bandwidth and Colocation sponsored by Easynews.com
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Re: [Asterisk-Users] RDNIS
Your telco doesn't supply RDNIS (aka OCN) by default. I have two PRIs, each from a different provider. Out-of-the-box, one PRI supports RNDIS/OCN and the other does not. I waiting for RDNIS/OCN to be enabled on the latter -- it was just a question of explaining what I wanted in the correct terms ie, explaining it's the OCN (orginial called number), isdn IE 115, that I wanted.On 9/7/05, Jonathan k. Creasy [EMAIL PROTECTED] wrote: Anyone had any success using RDNIS? I have a number (not on our PRI) being forwarded to another number (on our pri) by the CLEC. When I call the first number, it goes to the number on our PRIand if I understand correctly the RDNIS should be populated with the first number. It's empty though. Any ideas? -Jonathan ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Issue in calling mobiles....
Try sticking in an Answer() before dialing out on the Zap channel. On 8/24/05, Mauro Zanin [EMAIL PROTECTED] wrote: Hi dear group members, I have finally an Asterisk box working, capable of receiving and making calls. I have this issue while calling mobiles from our SIP softphones: -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] libpri mwi functionality?
Hi, Looking through the source code, I see two functions in the 'libpri' code for message-waiting: pri_mwi_activate, and pri_mwi_deactivate, but they are not used by anything anywhere. Is there a sample application somewhere which makes use of these functions? I believe that the ISDN MWI is used when a Norstar MICS has an external voicemail provider. I don't know exactly how this works yet, but if I had a dialplan command to use the pri_mwi functions, I'm sure I could figure it out :-) Thanks, -Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] libpri mwi functionality?
On 8/18/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote: I too am hoping to use these with the norstar MICS but I'm not 100% sure if it's that simple; the MICS seems to only want to allow that kind of featureset with MCDN (SL1 protocol) and I havne't had enough time to really play with that. I really am not very familiar with the Norstar family of PBXs I've managed to reprogram it for DIDs to each extension, OLI, and will risk minor changes to the call-routing, all to get nice integration with Asterisk. All I know for sure is what I've read in the Norstar Installer and Coordinator manuals, but that along with some forum posts on tek-tips.com lead me to believe it might work without the need for any extra equipment for my MICS. I guess I'll only know for sure once i can try it, but for that I need a Dialplan application like the SrxMWI, but for Digium PRI channels instead of Sirrix. Unfortunately, I'm not a good enough programmer to implement such a command on my own, but I'd be quite able to test, debug, and even suggest fixes if needed, once the basics functionality is there. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] yet another Asterisk and VMware question
On 8/12/05, Bruce Leetch [EMAIL PROTECTED] wrote: I've purchased a TDM11B card and have installed it in the box. Windows sees it as a PCI Simple Communications board. A Linux lspci doesn't show anything even vaguely resembling this card. This troubles me. You can run asterisk in a VMWare machine, but cannot use any hardware. VMWare simply doesn't give direct hardware access. Similar to video cards -- no matter what you have as physical hardware, your virtual machine always sees a standardized video device and for best performance, you use VMWare's drivers. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped between pstn norstar
I poured over my logs most of the morning. I'm fairly convinced at
this point the disconnect is coming from the Norstar 10 seconds after
the call was initiated. This points to the 'T310' timer, similar to
what is described here:
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094487.shtml
for the Cisco CallManager.
When comparing the log from a successful call to a failed call, I
noticed Asterisk was not passing back call-progress from the PSTN-span
to the PBX-span.
My zaptel.conf is the same is the nortel-asterisk-0.2.pdf by David
Gomillion except with only 2 spans instead of 4.
My dialplan is even simpler -- basically only a Dial(Zap/gX/${EXTEN})
in each context, where X is 1 or 2 as appropriate.
Attached are slightly edited logs (non-zap/pri stuff removed) for a
failed call. I also have the logs of a successful call if needed.
-Gary
On 8/11/05, Matt Fredrickson [EMAIL PROTECTED] wrote:
If you can post your `pri debug span X` (where X is the span in question)
we might be able to help you a little better.
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Protocol
Discriminator: Q.931 (8) len=36
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Call Ref: len= 2
(reference 5/0x5) (Originator)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Message type:
SETUP (5)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [04 03 80 90 a2]
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Bearer
Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]
Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]
Ext: 1 User information layer 1: u-Law (34)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [18 03 a1 83 82]
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Channel ID (len=
5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]
ChanSel: Reserved
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]
Ext: 1 Channel: 2 ]
Aug 10 16:13:37 VERBOSE[13685] logger.c: [28 09 b1 4f 50 45 4e 46 41 43 45]
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Display (len= 9)
Charset: 31 [ ]
Aug 10 16:13:37 VERBOSE[13685] logger.c: [70 08 80 35 37 31 32 36 33 36]
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Called Number
(len=10) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0)
'5712636' ]
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]-- Making new call
for cr 5
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]-- Processing
Q.931 Call Setup
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]-- Processing IE 4
(cs0, Bearer Capability)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]-- Processing IE
24 (cs0, Channel Identification)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]-- Processing IE
40 (cs0, Display)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]-- Processing IE
112 (cs0, Called Party Number)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Protocol
Discriminator: Q.931 (8) len=10
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Call Ref: len= 2
(reference 5/0x5) (Terminator)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Message type:
CALL PROCEEDING (2)
Aug 10 16:13:37 VERBOSE[13685] logger.c: [18 03 a9 83 82]
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0] Channel ID (len=
5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]
ChanSel: Reserved
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Aug 10 16:13:37 VERBOSE[13685] logger.c: [Span 1 D-Channel 0]
Ext: 1 Channel: 2 ]
Aug 10 16:13:37 VERBOSE[13685] logger.c: -- Accepting call from '' to
'5712636' on channel 0/2, span 2
Aug 10 16:13:37 DEBUG[13685] chan_zap.c: Enabled echo cancellation on channel 26
Aug 10 16:13:37 VERBOSE[15669] logger.c: -- Executing NoOp(Zap/26-1,
5712636 @ meridian callerid: dnid: 5712636 rdnis: ) in new stack
Aug 10 16:13:37 VERBOSE[15669] logger.c: -- Executing Dial(Zap/26-1,
Zap/g1/5712636) in new stack
Aug 10 16:13:37 VERBOSE[15669] logger.c: [Span 0 D-Channel 0]-- Making new call
for cr 32881
Aug 10 16:13:37 VERBOSE[15669] logger.c: -- Requested transfer capability:
0x00 - SPEECH
Aug 10 16:13:37 VERBOSE[15669] logger.c: [Span 0 D-Channel 0]
Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped betweenpstn norstar
On 8/11/05, Kris Boutilier [EMAIL PROTECTED] wrote: http://bugs.digium.com/view.php?id=4468 may have what you're looking for. If not, it sounds like a very similar problem. Likely you can work around it by issuing an Answer() to the Norstar before dialing out. That will satisfy the Norstars ISDN timers. Also, try enabling extended results on the Norstar - you should see a 'recov time exp' error message displayed on the originating set. Hope that helps. It does sound very familiar. I stuck an Answer before the Dial in the context for calls coming out of the Norstar, and can no longer reproduce the problem. Quick once-over of the pri debug logs shows the connections took longer than 10 seconds. Thanks for the help! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI dropped calls w/ asterisk dropped between pstn norstar
Hi, I dropped an asterisk server with a TE405P between a Norstar Meridian PBX and it's PRI PSTN connection. Everything seemed to work fine using a pass-thru-type dialplan configuration... except now we've realised that outbound calls to celphones get dropped upon connect, but not on every call (almost always the first try, but not the second). All other calls to non-celphones had no problems at all. I've had to reconnect the legacy PBX directly to the PSTN because it was too much of a problem. Asterisk was simply passing calls on one side to the other. The 'pri debug span' and regular logging doesn't appear have anything unusual or obvious. Limited testing could not reproduce the problem when calling from a SIP-phone out through the same PRI. The only things I can think of are not having 'transfer=no' anywhere in my zapata.conf (although none of the docs indicate this applies to a PRI), or having set the wrong switch type (set to dms100 when I maybe should have set national). I'm hoping someone else has encountered this and has a 'quick' fix or explanation, or can at least suggest what I should be looking for specifically in the pri debug output. Thanks, -Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF issues with SIPPhone?
On 8/8/05, Jason DiCioccio [EMAIL PROTECTED] wrote: I guess the problem is with SIPPhone then. I opened a ticket with them. I'll post their response when I have one. I wouldn't bet money on that yet... I've seen identical DTMF problems (doubled and mangled) digits and I've never used SIPPhone. I had traced it to a problem in the way asterisk handles out-of-sequence RFC2833 dtmf indications. See the -dev thread here: http://lists.digium.com/pipermail/asterisk-dev/2005-May/thread.html#12655 I tracked it down by using ethereal on the sip and rtp streams between my different (working and non-working) SIP providers. The only difference I could find was the order of the dtmf packets, and with the simple change to an 'if' statement mentioned in the -dev thread, the problems all went away. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] what is the problem with gmail and the list.
I do not think the problem with the lists since sometime July 29th was specific to Gmail... If you check the web archives, you'll see both regular posts and others (non-Gmail) asking about problems. My best guess is a subscriber's domain expired or some other similar problem which clogs mailserver queues. Hopefully the list admin will post an update once the problem is solved. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Simpletelecom dead?
Hmmm Can't place calls... Can't access website... Neither of the 3 nameservers answer anything... Anyone heard/know something to explain all this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HasNewVoicemail not being called if user hang up after leaving VM ??
When the person hangs up, Voicemail exits with -1 and you jump to extension 'h'.
Just create an 'h' extension which will get processed when the user
hangs up, like this:
exten = h,1,HasNewVoicemail(2002)
On 5/19/05, Mike Dent [EMAIL PROTECTED] wrote:
Hi,
it seems if a user leaves voicemail and hangs up the call when done, then
HasNewVoicemail never gets called on the next line in the context.
However if they press # to finish their VM, then it moves to
HasNewVoicemail and this
works?
eg:-
..
exten = 2002,3,VoiceMail(u${OFFICEVM})
exten = 2002,4,HasNewVoiceMail(2002)
..
exten = 2002,105, do something cos vm has been left.
I'm using 1.0.7
thanks
Mike
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Re: [Asterisk-Users] FREE music for downloading
On 5/18/05, Paul Mahler [EMAIL PROTECTED] wrote: http://www.signate.com/moh.php Would be better if you're site worked... HTTP/1.1 400 Bad Request Date: Thu, 19 May 2005 02:16:09 GMT P3P: policyref=http://p3p.yahoo.com/w3c/p3p.xml;, CP=CAO DSP COR CUR ADM DEV TAI PSA PSD IVAi IVDi CONi TELo OTPi OUR DELi SAMi OTRi UNRi PUBi IND PHY ONL UNI PUR FIN COM NAV INT DEM CNT STA POL HEA PRE GOV X-Host: p1w7.geo.scd.yahoo.com X-INKT-URI: http://us.geocities.com/server-errors/pd_bad_request.html X-INKT-SITE: http://us.geocities.com/server-errors Last-Modified: Fri, 31 Oct 2003 22:51:20 GMT ETag: 2980e-8c-3fa2e768 Accept-Ranges: bytes Content-Length: 282 Connection: close Content-Type: text/html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF asterisk-2-asterisk using SIP w/ dtmfmode=rfc2833
Hi, I'm am getting doubled DTMF on some digits with one of my providers who also uses asterisk. We're using SIP, with dtmfmode set to rfc2833, and the codec G.711. Once out of every five or ten calls, there are no problems, but more often than not, the DTMF is getting doubled-up on at least one of the digits of the extension dialed. I've tested with a CVS-HEAD from Febuary, and just now reproduced it with a fresh install (CVS-HEAD-05/16/05-20:22:32). I've used ethereal to capture the RTP stream and cannot see anything wrong with the 'RTP EVENT' packets -- whether my asterisk box sees the correct extension or not, the RTP EVENTs look okay as far as I can tell. There's no evidence of the DTMF being doubled before getting to my box. Has anyone else encountered this problem and found a solution? If the RTP EVENTs are good, where should I be looking for the problem? Thanks in advance for any help... Kinda stumped, -Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with MeetMe
You need to have the ztdummy kernel module loaded for timing purposes if you do not have any other zap devices. On 5/11/05, Daniel Salama [EMAIL PROTECTED] wrote: I'm trying to configure some meet me rooms in asterisk 1.0.2 and I'm getting the following problem: -- Executing MeetMe(SIP/3210-38a9, 0224|qM) in new stack == Parsing '/etc/asterisk/meetme.conf': Found May 11 14:05:46 WARNING[20050]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/pseudo': No such device or address May 11 14:05:46 ERROR[20050]: chan_zap.c:6687 chandup: Unable to dup channel: No such device or address May 11 14:05:46 WARNING[20050]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device May 11 14:05:46 WARNING[20050]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') I have the following in meetme.conf [rooms] conf = 0224 What could be happening? I don't have any digium cards on the machine. lsmod shows: Module Size Used byNot tainted zaptel182080 0 The other modules are not related to zaptel or asterisk. ls -l /dev/zap/ps* shows: crw-r--r--1 root root 196, 255 May 11 10:07 /dev/zap/pseudo Any ideas? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail
Same here... pratically all Asterisk mail now going to the Spam folder. I keep un-spamming them. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Livevoip DTMF via IAX almost
Umm... this is kinda interesting... IAX means Inter-Asterisk-Protocol, right? Apart from Asterisk and a few soft- hard- phones, what else uses IAX? In other words, what is LiveVOIP using on their end to do IAX? I'd assume one or more Asterisk servers.. So this is not like trying to get different implementations of SIP or any other protocol to work among different vendors it's one protocol by one 'vendor'. Does anyone know exactly what version of Asterisk/IAX they are running? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Passing varibles *out* of macros
Have you tried putting in some NoOp lines to verify the values of
${screenresult}?
Also, wouldn't you get the desired result by removing the 'g' option
from your Dial()?
You might want to add an 'h' extension for further processing on the
dead channel.
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Re: [Asterisk-Users] MeetMe2 admin functions
Am I supposed to create an admin and user menu context that I get sent to when I press * from the conference? That's what I decided to do after having similar problems and looking at the source. I only compared the source long enough to realize that the menu functions were coded differently and meetme2 seemed to have some functionality stripped out for dealing with DTMF tones. Grepping for 'DTMF' is the quickest way to find the menu code in either source file. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 800 Termination
What date was this? I've been waiting since Jan 24 on my 'pending' US50CA number -- I think you got VERY lucky! On Thu, 10 Mar 2005 23:59:00 -0700, Paul Fielding [EMAIL PROTECTED] wrote: Mine was up with LiveVoip within 30 minutes of ordering via the online website. And that was at midnight on a Saturday night. Of course, they don't guarantee that, I think I just got lucky... :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] country/city codes
Aren't country codes 1 or 2 digits? Area codes are 3 digits: 44 is a country code, 441 is an area code... the country code for Bermuda is '1', same as Canada, US, and most of the Carribean nations. Check this out for the solution you need: http://lists.digium.com/pipermail/asterisk-dev/2004-May/004151.html On Thu, 3 Mar 2005 06:25:09 -0800, VoIP Services [EMAIL PROTECTED] wrote: Some country codes are three digits long. Some are two. e.g. UK 44 , Bermuda 441 Does anyone know a formula for determining which part of a dialled number is the country code and city code ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x101p + Nortel ATA2
Hi, Does anyone have any experience connecting Asterisk to a Meridian system using an ATA2 and x101p? The basics work -- I can make outbound calls, receive inbound, and use flash to transfer calls, but certain things do not work, specifically with calls from internal extensions. - Does the Meridian/ATA2 pass any kind of callerid info? We do not have external callerid, but I'm not even getting extension numbers. - Calls involving an external line can use DTMF, but calls from another internal extension cannot. This is a problem for voicemail! I've tried *809 but it hasn't helped. Is this a limitation of the Meridian which won't pass DTMF internally? - Calls from internal extensions do not detect hangup properly, external calls are ok. So if an internal extension calls to leave a voicemail, the recording goes on until I do a sofft-hangup from the CLI. The plan was to use Asterisk as a voicemail server, but those three issues make this setup completely useless for that! Thanks, -Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] return code of app in dialplan
Hi, I'll probably kick myself when I read the replies to this... How do I test the return code of an app in the dialplan? I need to test if the app, MYSQL() in this case, returned -1 or 0. It's easy to see after-the-fact in the log output, but I need the result in the dialplan, I just can't find which variable stores the actual return-code. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
