Re: [Asterisk-Users] Aastra 9133i registration errors
I think the 9133 is like the 480i, in which you have multiple 'line' keys. I think its trying to register line 2 as well as 1, and you may not have anything set for line 2. If you make 2 the same as 1, incoming calls while you are on line one will roll-over to the line 2 key. If you don't want line 2, I'd remove any references to it in your MACADDRESS.cfg files. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Dave Cotton [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, November 19, 2005 4:57 AM Subject: Re: [Asterisk-Users] Aastra 9133i registration errors On Wed, 2005-11-16 at 20:17 +, Chris Bagnall wrote: Hi all, I have a pair of new Aastra 9133i phones here that I'm testing for receptionist duty at a couple of places next week and they don't seem to be registering with * correctly. I've set the phone up with the following entries in the appropriate tftp config file: sip line1 auth name: 205 sip line1 password: sip line1 user name: 205 sip line1 display name: 205 sip line1 screen name: Chris sip proxy ip: sip.minotaur.cc sip proxy port: 5060 sip registrar ip: sip.minotaur.cc sip registrar port: 5060 The error I'm getting in asterisk's console log is: NOTICE[8801]: chan_sip.c:9835 handle_request_register: Registration from 'No User sip:[EMAIL PROTECTED]:5060' failed for '10.10.1.251' - Wrong password However, the phone seems to be able to make and receive calls without any difficulty, and the phone appears correctly on a sip show peers. It seems to be purely cosmetic, but I'd love to know why the error's there. My first thought was that I probably don't need both sip proxy and sip registrar configured in the tftp config file, but I don't fully understand the difference between them in Aastra-speak (or which one * should be under). With my 9112i I only have registrar entries and have no such messages. But if I set qualify it returns unreachable so I just have qualify=no -- Dave Cotton [EMAIL PROTECTED] ___ --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 ___ --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] please recommend phones with adsi.
I sure like my Aastra 390, the voicemail ADSI app works pretty well (only a couple of incompleted functions, like not exiting by hanging up the speakerphone, rather than go to a reorder tone. As for the 'look at the wiki' comment, I'm not trying to get on anyone's badside, but Dmitry was asking for recommendations, not documentation. Sorry, not pointing fingers, but I see that 'blanket answer' of going to to Wiki all too often on here lately ;) If I was asking for recommedations of good jazz music, it wouldnt help me to have someone tell me to go to Sam Goody. Chris - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 27, 2005 7:53 AM Subject: Re: [Asterisk-Users] please recommend phones with adsi. Look on the wiki which is located at: http://www.voip-info.org/ On 10/27/05, Dmytro Mishchenko [EMAIL PROTECTED] wrote: Hello, can somebody recommend me any hard or may be even softphones which support ADSI. I would like to work with Asterisk voicemail application using ADSI. Thanks, Dmitry ___ --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 ___ --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 ___ --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] please recommend phones with adsi.
So much for not stepping on toes. Incidently, there have been dev-asterisk posts in the past relating to ADSI tones being processed through a SIP channel, so theoretically, a softphone 'could' exist. I've been hard-pressed just to find any documentation via google explaining any ADSI-command-standards-list. I've seen some ADSI telephones that have little more than glorified caller ID, while others (like the 390 or 450) that have priorities, soft keys, etc etc) So, yes, it is also possible that a softphone could exist Chris - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 27, 2005 12:07 PM Subject: Re: [Asterisk-Users] please recommend phones with adsi. On 10/27/05, Chris Coulthurst [EMAIL PROTECTED] wrote: I sure like my Aastra 390, the voicemail ADSI app works pretty well (only a couple of incompleted functions, like not exiting by hanging up the speakerphone, rather than go to a reorder tone. As for the 'look at the wiki' comment, I'm not trying to get on anyone's badside, but Dmitry was asking for recommendations, not documentation. Sorry, not pointing fingers, but I see that 'blanket answer' of going to to Wiki all too often on here lately ;) I hope you are talking about the same post from Dmitry. The following is Dmitrys post: Hello, can somebody recommend me any hard or may be even softphones which support ADSI. I would like to work with Asterisk voicemail application using ADSI. Thanks, Dmitry For which I responded look at the wiki and I even included a direct link to the wiki. I don't see in any way how this doesn't answer his question. In any case it sure doesn't look like he has any clue of *any* ADSI phones, since he asking for a softphone that supports ADSI, while I can't say it doens't exist or there is no use for it (BTW, I'm almost sure it doesn't exist), there sure isn't a market for it. Since ADSI is something made to work on Analog networks. While if you were asking for good jazz music the Sam Goody answer wouldn't do, if you had no clue what jazz music is then the Sam Goody answer is the right answer. In most cases when you see that blanket answer of go to the wiki, it is becuase the person posting the question has thru the question told everyone I havn't seen the wiki yet. Which BTW was the case here. Hope this helps you understand why that answer was in place. Please don't take this as being on my badside I'm just trying to explain to you what RTFM means. All of us are busy with somethings, we take our time to answer the questions here on the list it doesn't mean that we are here to do the work for you so that you could be a lazy bum. If someone is lacking the knowledge of searching the wiki and shows that thru posting that question of any soft phones supporting ADSI, I answerd the question with the most repect I could gather for the 2 seconds by directing them to the wiki, since with that question they showed they had no clue the wiki exists, and if Dmitry will tell me that he did know about the wiki and still posted the question the way he did, then he did not deserve my 2 seconds. ___ --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 ___ --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] Taking the plung to CVS HEAD
If your 1.0.9 install is (on the /usr/src/asterisk tree) complete, you might unpack the CVS source somewhere else other than /usr/src (maybe /usr/local/src or /usr/src/cvs). Most importantly, PLAN AHEAD. It would seem that the more Asterisk evolves, the less-tolerant it is natually becoming to obsoleted methods and code. Little things like commas where pipes should be could be a pain. Also, think to make sure you didn't install, and depend on a patch (spandsp for example, the patches are very version-dependant). Copy /etc/asterisk, /var/lib/asterisk, /var/spool/asterisk and /usr/lib/asterisk to a backup location FIRST. I remember when I upgraded to CVS-HEAD back in mid August. I learned the hard way to triple-check the little things. If you copy all the files (maybe a restore script would be handy), you should be able to just run a make install from each source-tree's directory to return it to normal. Oh, don't forget to erase the contents of /usr/lib/asterisk/modules before you 'make install' new ones. Chris - Original Message - From: Dave Grey [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 27, 2005 7:01 PM Subject: Re: [Asterisk-Users] Taking the plung to CVS HEAD On Oct 27, 2005, at 9:52 PM, Eric Bishop wrote: We are running 1.0.9 STABLE on all of our machines. I am about try and upgrade one machine to CVS HEAD as all this echo cancellation improvements sound enticing. Can anyone recommend a) A procedure to cleanly upgrade from STABLE to HEAD b) A procedure to ensure I can back out and go back to 1.0.9 easily I have looked on the wiki but couldn't find much about this. ___ --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 The easiest thing, I'd think, would be just to install the HEAD with an entirely different basedir and don't touch your existing install at all. That way it is a simply matter of running a different executable to switch between the old and the new. That's what I did here, at any rate, and it worked nicely, but my living-room is certainly not a production environment. Stay tuned for possible contradictions from the pros. lyd ___ --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 ___ --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 MWI
I think I have an idea of what dto do here. Look in your sip.cfg file for a line starting with MSG_WAITING under the CALLPROGRESS section. It defines the tone chirp you hear for message waiting notification. I'll bet if you zero out the values it would stop alerting you. P.S. It might be in ipmid.cfg if you have that file instead Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Wilson Pickett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 17, 2005 12:33 AM Subject: [Asterisk-Users] Polycom MWI Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips? ___ --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 ___ --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 MWI
I, myself would be happy if you could configure the DND button to send back to the server, shall we say, more configurable options. Instead of just having it option to indicate busy on DND active, it would be nice to enlist another state, maybe still allow calls through (DND override) if a specific alert-info indication were present. I, myself, use Polycom 5xx-series phones throughout my home and office. It would be nice to hit that 'leave-me-alone-key' so no callers get through, but I could still allow certain extensions or whitelisted Caller IDs get through. I know this can be done in the asterisk dialplan, but it defeats the purpose of having that nearly useless, unreassignable DND button on the phone! Take care, polycom config files become less scary the more you read through em! Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Wilson Pickett [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 17, 2005 11:46 AM Subject: Re: [Asterisk-Users] Polycom MWI MESSAGE_WAITING se.pat.misc.1.name=message waiting se.pat.misc.1.inst.1.type=silence se.pat.misc.1.inst.1.value=1 se.pat.misc.1.inst.2.type=silence se.pat.misc.1.inst.2.value=2 se.pat.misc.1.inst.3.type=silence se.pat.misc.1.inst.3.value=1/ I didn't bother taking out the unnecessary stuff, I just changed where it said chord to silence, this way if I needed to bring it back I could just change silence back to chord. Thanks guys. I will be using config files RealSoonNow but for the moment I haven't gotten to it. I certainly will have to if I get more phones though :) I wish this was at least a setting option on the phone itself, better yet the DND button could silence it logically, but noo. ___ --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 ___ --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] Teliax users, g729 question
FYI, Im using g729 with Teliax, and have been for about 1 week with no problems, good audio quality. They DO seem to drop registrations unexpectedly at times, but as for codec usage, so far so good. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-users-list asterisk-users@lists.digium.com Sent: Monday, October 10, 2005 6:58 PM Subject: Re: [Asterisk-Users] Teliax users, g729 question I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. show translations verifies that the registration took place. When I place a call, having allow=g729 as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a call with g726 as the only allow line, then the call completes as desired. I realize I could just use that (g726), but it seems odd that I cannot connect using g729 when both of us support it. I have checked, double checked, triple checked my account with teliax, and made sure that the g729 box is checked for both sip and iax. I have also contacted support, they have responded, but not with anything that would be considered helpful. My question then to you all is this: Are you connecting to Teliax via g729? if so, how... what are you doing that I might be missing? Your guidance will be most appreciated. I just tried g729 with teliax this morning. It worked fine in both directions using three different did's from them. I did have one test call where audio was one way though. During those test calls I watched the CLI and the calls definitely were g729 without a doubt. I used the exact same teliax server you show above. I'm running cvs-head from yesterday. If you do a iax2 debug, you should be able to spot which system is not compat with g729. That should lead you into diagnosing the problem a little deeper. Rich, Thanks for taking the time to check and respond, and for your advise. I did as you said, and here is the output: pbx*CLI iax2 debug IAX2 Debugging Enabled Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00016ms SCall: 2 DCall: 0 [208.139.204.228:4569] VERSION : 2 CALLED NUMBER : 1917 (I XXX out the number) CODEC_PREFS : (g729) CALLING NUMBER : 4401 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: John Reynolds LANGUAGE: en USERNAME: reynj FORMAT : 256 CAPABILITY : 63744 ADSICPE : 2 DATE TIME : 189432385 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 00073 DCall: 2 [208.139.204.228:4569] AUTHMETHODS : 2 CHALLENGE : 303570605 USERNAME: reynj Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00080ms SCall: 2 DCall: 00073 [208.139.204.228:4569] MD5 RESULT : 6006ef53daa295d58f5292837a4edb60 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT Timestamp: 00060ms SCall: 00073 DCall: 2 [208.139.204.228:4569] CAUSE : Unable to negotiate codec CAUSE CODE : 58 Oct 10 16:18:03 WARNING[364]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec = I am I wrong in thinking that I my call is trying to connect via g729, but that Teliax is rejecting it? Okay, in the above, you transmit a packet that has FORMAT : 256 which is g729 (see 'show codecs'). After the challenge/response, they send you 'REJECT, Unable to negotiate codec'. So, yes, teliax is rejecting your call due to the lack of g729 support on their side. I was just going to log into their web site to check my settings, and their web server is broken right now. I've also been experiencing many errors with teliax.com in the form of no call progress when dialing any call. So, they are either having serious problems, went broke, or something. On the g729 issue, the folks at teliax.com would have had to either purchase codecs from digium, or, installed an illegal codec. Presumably, if they installed xx number of g729 licenses and those licenses were already consumed, would they reject the next call (like above)? Probably. So, can't tell if they simply ran out of licenses, have a configuration problem for your account, or their system is simply broken. In any case, you can't do anything to fix it other then to open a problem case with them or use a different codec. Gsm is pretty good. :) Rich ___ --Bandwidth and Colocation
Re: [Asterisk-Users] Teliax users, g729 question
Make sure you have g729 turned on from the Teliax customer panel on their website. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; John Reynolds [EMAIL PROTECTED] Sent: Saturday, October 08, 2005 8:59 AM Subject: Re: [Asterisk-Users] Teliax users, g729 question I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. show translations verifies that the registration took place. When I place a call, having allow=g729 as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a call with g726 as the only allow line, then the call completes as desired. I realize I could just use that (g726), but it seems odd that I cannot connect using g729 when both of us support it. I have checked, double checked, triple checked my account with teliax, and made sure that the g729 box is checked for both sip and iax. I have also contacted support, they have responded, but not with anything that would be considered helpful. My question then to you all is this: Are you connecting to Teliax via g729? if so, how... what are you doing that I might be missing? Your guidance will be most appreciated. I just tried g729 with teliax this morning. It worked fine in both directions using three different did's from them. I did have one test call where audio was one way though. During those test calls I watched the CLI and the calls definitely were g729 without a doubt. I used the exact same teliax server you show above. I'm running cvs-head from yesterday. If you do a iax2 debug, you should be able to spot which system is not compat with g729. That should lead you into diagnosing the problem a little deeper. ___ --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 ___ --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] Variable for codec used?
Is there an easy (or even a hard) way to save to the CDR a userfield value with the call's codec in it? Chris Coulthurst [EMAIL PROTECTED] ___ --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] 911 Q
I installed a Marquee sign (aka reader board), which was sent emergency information via an RS-232 serial port. It was pretty nifty, as it was during to 'everywhere must have caller ID' phase in the 90s. Most signs are cheap, and can just be placed in the clubhouse window. You could even have nice littlearrows pointing the direction of the 911 caller's dwelling... Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Joel Newkirk [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, September 30, 2005 7:20 AM Subject: [Asterisk-Users] 911 Q OK, got a question on 911. Looking into setting up a couple asterisk servers at a country club, with VOIP phones in each of 100 short-term residential rental units. Approx 100 extensions, approx 24 outside lines. Since everything is geographically at one location, reaching 911 correctly shouldn't present a problem. However, the club wishes to ensure that 911 authorities are able to identify the precise rental unit placing the call. How can we achieve this, short of 'reciting' the unit number aloud at the beginning of the placed call? Thanks for any advice/tips. j ___ --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 ___ --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] SoundPoint IP Attendant Console
So the IP 601 is the 600 with a few extras? Looks like Polycom dropped the ball again -- yet another pretty phone with NO BACK LIGHT. Does the design team at Polycom have their brains unscrewed? I've been playing with some Aastra phones lately, with limited success on working properly. The only motivations for checking out these other phones were the two things that polycom has been lacking from the get-go...REAL LEDs and Backlit displays. PBX and KSU telephones have had backlights for more than a decade now. If I wanted a phone I can't see in the dark, I'd use the Panasonic D1232 I already had... I've been using a set of 300s, 500s and one 600 for 4 months now, and have been very happy with the results. As soon as Polycom pulls its head out of its you-know-what and makes LEDs, backlights, and maybe a usable administrator's guide, I'll consider them for my customers in the future. So far, looks like Aastra is winning the bid...(barely) Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 21, 2005 6:36 PM Subject: Re: [Asterisk-Users] SoundPoint IP Attendant Console Bartosz Jozwiak wrote: Does anybody use SoundPoint IP Attendant Console for Polycom IP 601 with asterisk ? Is it going to work with hints in dial plan ? Since it is not even shipping yet (it was just announced two days ago), the answer is no. However, we have had a test unit for some time (and we have one in our booth at VON), and yes, it works just like the built-in buttons on the phone. The only issue today with displaying hint status is an artificial limit of eight (8) 'buddies' in the Contact Directory to watch. Once Polycom has released the final firmware for the phone with support for a larger number of watched contacts, the expansion module will be fully usable with Asterisk. ___ --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 ___ --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] Voicemail
Try changing the Gecos field in /etc/passwd for the user Asterisk is running
as. This is normally the 5th field, and is used for user information. I
made mine just say PBX and now it says [EMAIL PROTECTED] on text
messages/pages. This is from a default CentOS install with sendmail
conventionality.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Damon Estep [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 19, 2005 6:55 AM
Subject: Re: [Asterisk-Users] Voicemail
AHHH HA! Yes that answered all of my questions.
Ok.. everything is right with the world, except I still can't get the
'pager' e-mail address to not be
[EMAIL PROTECTED]
I have a serveremail= string and that gets set for e-mails.. but I
don't see anything about a pager string. I set the pagerfromstring
but that also seems to not change the e-mail address used for pagers.
On 9/18/05, Damon Estep [EMAIL PROTECTED] wrote:
Download cvs head and look at
/usr/src/asterisk/configs/voicemail.conf.sample
All of the variables for email, page, etc are listed in the sample
files, it is more comprehensive than many of the other samples.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt
Sent: Sunday, September 18, 2005 6:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail
While we are on the subject.. how do you modify the TXT message that
gets send to the 'pager'... Is that hard coded.. or can that be
changed?No variables I change in voicemail.conf seem to change the
from address, etc.
On 9/18/05, Damon Estep [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, September 18, 2005 8:32 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voicemail
When I receive a voicemail notify via e-mail I would like receive
not
the sender phone-number, but the sender name. Where can I configure
this and how? Is it possible to have some example?
Thank
Luca
This will do it (in voicemail.conf) but I think the default does as
well. Are you actually getting caller ID name delivered when a call
comes it?
emailbody=${VM_NAME} ${VM_MAILBOX}\n\nYou have a new voicemail
message
from ${VM_CALLERID}. The message is ${VM_DUR} long and was left on
${VM_DATE}.
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Re: [Asterisk-Users] Voicemail
Oh btw: here's my voicemail.conf line for just getting the number, not the
name:
emailbody=Fm:${VM_CIDNUM}\n${VM_DATE}\nDur:${VM_DUR}\n
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Matt [EMAIL PROTECTED]
To: Damon Estep [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 19, 2005 6:55 AM
Subject: Re: [Asterisk-Users] Voicemail
AHHH HA! Yes that answered all of my questions.
Ok.. everything is right with the world, except I still can't get the
'pager' e-mail address to not be
[EMAIL PROTECTED]
I have a serveremail= string and that gets set for e-mails.. but I
don't see anything about a pager string. I set the pagerfromstring
but that also seems to not change the e-mail address used for pagers.
On 9/18/05, Damon Estep [EMAIL PROTECTED] wrote:
Download cvs head and look at
/usr/src/asterisk/configs/voicemail.conf.sample
All of the variables for email, page, etc are listed in the sample
files, it is more comprehensive than many of the other samples.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt
Sent: Sunday, September 18, 2005 6:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail
While we are on the subject.. how do you modify the TXT message that
gets send to the 'pager'... Is that hard coded.. or can that be
changed?No variables I change in voicemail.conf seem to change the
from address, etc.
On 9/18/05, Damon Estep [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, September 18, 2005 8:32 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voicemail
When I receive a voicemail notify via e-mail I would like receive
not
the sender phone-number, but the sender name. Where can I configure
this and how? Is it possible to have some example?
Thank
Luca
This will do it (in voicemail.conf) but I think the default does as
well. Are you actually getting caller ID name delivered when a call
comes it?
emailbody=${VM_NAME} ${VM_MAILBOX}\n\nYou have a new voicemail
message
from ${VM_CALLERID}. The message is ${VM_DUR} long and was left on
${VM_DATE}.
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[Asterisk-Users] Firmware upgrade Aastra 480i CT
Does anyone have success in upgrading Aastra/Sayson 480i CT firmware? All I get, no matter what I've tried is "Unable to upgrade firmware". tftpd is working because the dialplan freshens, and I have aastra.cfg whatevermacaddressfile.cfg and firmware.st in /tftpboot Am I missing something stupid? Is there another way to upgrade it? Chris Coulthurst [EMAIL PROTECTED] ___ --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] Anyone using Telasip, Caller ID presentation outbound??
II noticed that Caller ID presentation is not making it to my cell phone through outound Telasip calls and I don't know why. It may very well have been this way for awhile (or always, not sure I called my cell phone during telasip testing). Does Telasip expect a different format than SetCIDNum(NXXNXX) ? It hasalways worked for the Teliax lines. BUT--- It doesn't have a problem making it to landline phones Ive tried... I user Verizon for the cell and Qwest for my incoming analog (with callerID) lines... Chris Coulthurst [EMAIL PROTECTED] ___ --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] How to execute StopPlayTones when a SIP phone is answered
I'm trying to find a way to generate an 'internal
extensions' tonelist but I can't seem to find anything on how to do this.
My idea was to start a Playtones(intercom) tonelist and not indicate ringing to
the line (dead air). But then, somehow StopPlayTones needs to be run once
the ringing telephone picks up.
This seems like a dirty way to do this. I
envision an option to the Dial cmd's option 'r', where you could specify a
ringtone to play if not the default, i.e.
In indications.conf:
[us]
...
...
ring =
400+450/400,0/200,400+450/400,0/2000intercom =
400+450/400,0/200,400+450/400,0/2000 ;FRESHLY ADDED AND STOLEN FROM [uk]
section.
1001,1,Dial(SIP/1001,20,r{intercom})
For what its worth, I'm trying to use the standard
UK ringtones for an internal extension. This behavior mimics several
different PBXs and KSUs on the market.
Does anyone have something like this
working?
Chris Coulthurst
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Overhead Paging Systems...
The two most common companies to make paging equipment are Viking and Bogen. Bogen even resells ATAs for paging now. http://www.bogen.com or http://www.vikingelectronics.com Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, September 01, 2005 7:33 AM Subject: [Asterisk-Users] Overhead Paging Systems... Hey all, I know you all saw the topic and let out a groan. However, I understand how to get an overhead paging system to work with respect *, however I am now looking for a small(?) paging amp, that I could hook 3 or 4 horns to. I would like to just have the * extention be routed to a soundcard and out an output, so I would like an amp that is voice signal activated. Has anyone found anything like this? This is my first * installation, and I havn't been finding too much on google that helps me. ~kurth ___ --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 ___ --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] teliax
Yes ever since the hurricane hit, I have had crackling on the line and MAJOR delays and even some echo. Some odd pings to Teliax have been noted as well. I have had no problems with Telasip (my backup). But on a similar note, I have tried to dial many texas, mississipi and florida phone and fax numbers in the last two days, and even on POTS lines I get fast busys, static or otherwise incomplete calls. Its to be expected that the network in general is one big re-route right now, so these kinds of problems shouldn't be unexpected, especially with IP calls if certain hubs are rerouting through less-capable alternates. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Chris [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 30, 2005 6:31 AM Subject: Re: [Asterisk-Users] teliax No, I don't have service with them. I am thinking about getting service from them and I had some specific questions about porting telephone numbers and clear up some things about their packages. Regards, Chris - Original Message - From: Rick Baranowski [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, August 29, 2005 4:59 PM Subject: RE: [Asterisk-Users] teliax I have had some calls from time to time that crackle and or cut out. Is this what is happen to you Chris? They don't happen to the calls to and from the * box just on outgoing call that go through Teliax. We have the box in a data center that is offsite so in a sense if we where having a connection problem one would think that we would have an issue calling to VM, ext's, etc. We use them to Terminate all of are calls so it should not be an issue on our end. We have a low call volume and I have not had the time to trace this down. I have talked to Teliax about it but they say they are not having any issues. Am I alone on this? If not, can you guys chime in so we can help trace this down. Thanks Rick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Coulthurst Sent: Monday, August 29, 2005 1:40 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] teliax What is the problem you are querying about anyway? I've noticed some VERY bad audio on the circuit when I initiate calls. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Chris Mason (Lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 29, 2005 11:09 AM Subject: Re: [Asterisk-Users] teliax Chris wrote: Is there a problem at Teliax? I'm looking for a VoIP provider and when I call them they never answer the phone and the voice mail says it's full. I don;t see any network problems, and I monitor Teliax and a few other providers. Teliax is my main provider and I have never had any problems worth worrying about. Send them an email, I find they always respond, at least if the question is reasonable. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --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 ___ --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 ___ --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 ___ --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 ___ --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
Re: [Asterisk-Users] teliax
What is the problem you are querying about anyway? I've noticed some VERY bad audio on the circuit when I initiate calls. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Chris Mason (Lists) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 29, 2005 11:09 AM Subject: Re: [Asterisk-Users] teliax Chris wrote: Is there a problem at Teliax? I'm looking for a VoIP provider and when I call them they never answer the phone and the voice mail says it's full. I don;t see any network problems, and I monitor Teliax and a few other providers. Teliax is my main provider and I have never had any problems worth worrying about. Send them an email, I find they always respond, at least if the question is reasonable. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --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 ___ --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] All Page ??
Polycom most definately should add a provision for off hook voice announcing. It would be nice to have a receptionist announce a call for you over the speakerphone while you are using the handset. I don't see why it would be anything more than a programming limitation, but then it becomes an issue of getting asterisk to talk to it. Even still, I'm sure the alert-info flag would be usable here as well, since you can already control not only ringer types, but call waiting cadences as well. Is there a 'suggestion box' for upcoming SIP software releases on Polycom's site? Or maybe someone knows the right person to drop the hints to in the company? In the meantime, your best bet is really overhead paging amplifiers, which you can get from Viking that work over Zap channels quite well. It can even integrate with call park announcing. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Matthew T. O'Connor matthew@zeut.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 21, 2005 7:42 PM Subject: Re: [Asterisk-Users] All Page ?? Steve Maroney wrote: Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. I'm looking for an answer to this problem also. I am putting an Asterisk system into our new office. In our old office we used the old phone system to act as an intercom, you hit page all and your voice comes out of the speaker on several handsets throughout the office. This allows you to announce information or to the whole office, simply announcing to someones desk doesn't work since our people move around a lot and are not always at their desk. Anyway, I have some Polycom phones, and I have Autoanswer working with Asterisk, but which ever phone happens to answer the call first is the only one who's speaker my voice comes out of. Anyone have an answer to this problem? Thanks, Matt ___ 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] Transferring from cell phone Revisited
Same problem as before. The features.conf has the default settings of *1
the this and *2 for that, and the # for transfers. They are uncommented.
They do not work for me.
And if this is something I'm doing wrong (I hope), I would really prefer a
double-# strike in quick action, since so many things including voicemail
like the pound key. Since *1 is a pre-set option, I assume that you can do
this, as well as set the interdigit requirement?
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-08-15 18:11:51 UTC
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Chris Coulthurst [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 16, 2005 5:01 PM
Subject: Re: [Asterisk-Users] Transferring from cell phone
Its left as default, and when I press the # nothing happens, but the
remote caller doesn't hear the DTMF tone.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Michiel van Baak [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, August 16, 2005 10:05 AM
Subject: Re: [Asterisk-Users] Transferring from cell phone
On 22:31, Mon 15 Aug 05, Chris Coulthurst wrote:
I set up a context to allow me to call in to my * server (via Teliax in
this case using IAX2) from my cellphone, and let me do a number of
things, including dial other extensions, AND dial outbound again so
callers could see my proper work CallerID when I use this service. Is
there a way to be able to transfer calls to other extensions of my
asterisk server FROM the cell phone/ This isn't a Zap channel, so I'm a
bit lost, but did specify the 'T' option in dial. Here's my context.
Is this possible to do??
[aa_chris_disa]
exten = s,1,Read(DIALNUM,custom/enter-num-then-pound,21)
exten = s,2,Playback(connecting)
exten = s,3,GotoIf($[${LEN(${DIALNUM})} 5 ]?4:8) ; IF SHORTED THAN 5,
its internal so dial internal
exten = s,4,SetCallerID(Chris Mobile 205)
exten = s,5,Dial(Local/[EMAIL PROTECTED]/n) ;DIAL INTERNAL EXTENSION
exten = s,6,Playback(call-terminated)
exten = s,7,Goto(aa_chris_start,s,1)
exten = s,8,Gotoif($[${LEN(${DIALNUM})} = 7]?s,9:s,14) ;IF 7 DIGITS
DIALED, ITS LOCAL, PREPEND THE AREA CODE
exten = s,9,SetCIDNum(99)
exten = s,10,Dial(${IPTRUNK}/360${DIALNUM},,T)
exten = s,11,Dial(SIP/[EMAIL PROTECTED],,T)
exten = s,12,Playback(all-circuits-busy-now)
exten = s,13,Goto(aa_chris_start,s,1)
exten = s,14,SetCIDNum(99) ;NUMBER ISNT 7, OR LESS THAN 5 SO
AREA CODE WAS ADDED
exten = s,15,Dial(${IPTRUNK}/${DIALNUM},,T)
exten = s,16,Dial(SIP/[EMAIL PROTECTED],,T)
exten = s,17,Playback(all-circuits-busy-now)
exten = s,18,Goto(aa_chris_start,s,1)
exten = i,1,Goto(aa_chris_start,s,1)
Did you modify features.conf ?
If not, what happens if you puch the # key on your cellphone
when connected to * ?
If you did change it, try the key you configured there for
transfer :)
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
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[Asterisk-Users] Dial, RING with a digit interrupt
This is a new post, but its really a three-time retread. I hope someone has a clue on this, as it could be helpful in many circumstances: I am looking for a way to dial 'special'an extension (in house, like 102), whichare all Polycom IP. I'd like to ring the extension as normal, but have the option of, while the line is ringing, to press a digit, hop out to a new context and/or priority and do something else. Even a key system as basic as the Panasonic 616 let you press 1 while ringing an extension, and user-permissions allowing, would force the call in to off-hook voice announce. If you called an extension that was busy, pressing 6 would make the called party's phone ring you back when they hang up. These are two seemingly simple ideas, but I've not yet come up with a good concept of how. I've looked at the command RetryDial, which in many ways comes close, but misses. Something tells me this is going to be a "playtones" type of situation, but I have no clue if, or how its possible to generate ringtone on the handset, then drop in to the next priority to actually RING the extension, all the while waiting for a digit. Once it gets that digit, it has to stop the playtones, hangup the channel to clear it, and immediately ring it back with ALERT_INFO=answer/ring-answer. If I could make that happen, I swear I'll fart sunshine-dust. ;) Any ideas to move in the right direction, hints and similar suggestions are more than welcome! Chris Coulthurst [EMAIL PROTECTED] ___ 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] How to change RINGING style for internal calls
I'd like to have the ringing a caller hears to be more like a 'british' ring when I am calling an internal extension. The phones I'm calling already do this, now I'd like to find a way to make the same thing happen for the caller who waits... Any ideas? Chris Coulthurst [EMAIL PROTECTED] ___ 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] Voicemail file permissions
If you want to use samba for their checking, you COULD use smb.conf to control which users get in to which folders. You might make /var/spool/asterisk/voicemail/default readable by all, and make [username] entries in smb.conf to add 'admin user' accesses to only their voicemail directory tree. I know it is doable, but I can't remember all the commands in samba off-hand to do it. Even if this is not EXACTLY the best way to do it, at least it gives you a couple ideas to manipulate... One thing you might think about is using a script to map voicemail extensions to known samba usernames/groups, and have the script chown the specific directories to that samba username and group. I don't know what variables are passed (if any) from Comedian mail, but it doesn't really matter, since that script could read a list of ALL vm/username mappings each time voicemail is left, checked, deleted, and make a change to all of the vm users at the same time. Some would be redundant, but its not exactly eating up a ton of cpu cycles to chown users. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: hugolivude [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 17, 2005 5:56 PM Subject: Re: [Asterisk-Users] Voicemail file permissions Sorry, not following you at all. Perhaps it's a lack of Linux knowledge. I understood that Asterisk puts voicemails in var/spool/asterisk/voicemail/mailbox/###/INBOX. Are you suggesting that I could configure that to be a network drive on each user's computer? If so wouldn't that introduce all kinds of complications like what to do when their computer is off? Thanks for taking the time to respond though, most appreciated. Howard On 8/17/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Wed, Aug 17, 2005 at 07:48:29AM -0400, hugolivude wrote: Is there a way around this w/o giving everyone root privileges! Do you want to allow every user to delete another user's voicemail? If not, how do you sync voicemail users and samba users? I want each user to see, read and write (delete) their own voicemail ONLY (i.e. a user shouldn't be able to listen to someone elses voicemails). I gave each user an account on the Asterisk box and limited their access to their mailbox folder only. So don't waste your time on saving the voicemail on Asterisk. Save it on a specific folder in an imap server on the user's home directory. If you use a decent mail client, getting notifications for new mails in that folders, deleting them, playing them, and whatever should be easy. On the Asterisk side you only need to keep voicemail config in sync. Maybe it would be easier to just forward every mailbox nnn to [EMAIL PROTECTED] and use an aliases file to do the real forwarding. That way you keep the emails away from Asterisk's config. The downside: no message-waiting indicator. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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 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] DID on TDM400P Question?
I could see something working only if the telco was able/willing to set up a circuit that sends dtmf digits immediately after a circuit answers (kind of like some analog voicemail systems do on FXO ports). The context upon answering would have to take the digits and dialout accordingly to the right devices. This, however, would require two miracles, an act of congress, and a note to the telco from santa claus. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 17, 2005 8:59 PM Subject: Re: [Asterisk-Users] DID on TDM400P Question? Does anyone know if the current TDM400 card can take DID digits from the LEC? If so is there any reference to how to set this all up? As I get my current service from my LEC over an IAD, so would be sweet to just have trunks, not each channel specific to a number. Also if the above is possible, if the line is being used for DID, then is this only workable for inbound, or can I also seize the line and use it for outbound calls. I know with PRI's that is easy, but never had to play with this on an analog port level. Just having a PRI at home isn't practical, so not something I can really do. Not likely to work. Part of the reason is the TDM card essentially answers a call when ringing occurs. That answer essentially closes the tip-ring loop to the pstn (central office), and the central office will interpret that as a change in line status. The central office will not forward any more digits (dial pulse or dtmf) when that occurs. If the TDM card supported EM signaling, then one would have the signaling structure (to the central office) to support the wanted DID function; but that doesn't exist and the TDM chipset doesn't support EM. Last, the asterisk code would need to change in such a way as to listen for additional dtmf digits _after_ the TDM card closed the tip-ring loop, and I believe that code doesn't exist either. (Even if the code did exist, then what method would be used to signify to the central office that a call was answered by an asterisk phone?) Bottom line... not with a TDM card. ___ 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] Transferring from cell phone
Its left as default, and when I press the # nothing happens, but the remote
caller doesn't hear the DTMF tone.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Michiel van Baak [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, August 16, 2005 10:05 AM
Subject: Re: [Asterisk-Users] Transferring from cell phone
On 22:31, Mon 15 Aug 05, Chris Coulthurst wrote:
I set up a context to allow me to call in to my * server (via Teliax in
this case using IAX2) from my cellphone, and let me do a number of
things, including dial other extensions, AND dial outbound again so
callers could see my proper work CallerID when I use this service. Is
there a way to be able to transfer calls to other extensions of my
asterisk server FROM the cell phone/ This isn't a Zap channel, so I'm a
bit lost, but did specify the 'T' option in dial. Here's my context. Is
this possible to do??
[aa_chris_disa]
exten = s,1,Read(DIALNUM,custom/enter-num-then-pound,21)
exten = s,2,Playback(connecting)
exten = s,3,GotoIf($[${LEN(${DIALNUM})} 5 ]?4:8) ; IF SHORTED THAN 5,
its internal so dial internal
exten = s,4,SetCallerID(Chris Mobile 205)
exten = s,5,Dial(Local/[EMAIL PROTECTED]/n) ;DIAL INTERNAL EXTENSION
exten = s,6,Playback(call-terminated)
exten = s,7,Goto(aa_chris_start,s,1)
exten = s,8,Gotoif($[${LEN(${DIALNUM})} = 7]?s,9:s,14) ;IF 7 DIGITS
DIALED, ITS LOCAL, PREPEND THE AREA CODE
exten = s,9,SetCIDNum(99)
exten = s,10,Dial(${IPTRUNK}/360${DIALNUM},,T)
exten = s,11,Dial(SIP/[EMAIL PROTECTED],,T)
exten = s,12,Playback(all-circuits-busy-now)
exten = s,13,Goto(aa_chris_start,s,1)
exten = s,14,SetCIDNum(99) ;NUMBER ISNT 7, OR LESS THAN 5 SO
AREA CODE WAS ADDED
exten = s,15,Dial(${IPTRUNK}/${DIALNUM},,T)
exten = s,16,Dial(SIP/[EMAIL PROTECTED],,T)
exten = s,17,Playback(all-circuits-busy-now)
exten = s,18,Goto(aa_chris_start,s,1)
exten = i,1,Goto(aa_chris_start,s,1)
Did you modify features.conf ?
If not, what happens if you puch the # key on your cellphone
when connected to * ?
If you did change it, try the key you configured there for
transfer :)
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
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Re: [Asterisk-Users] Voicemail file permissions
My suggestion would be, use the externnotify=/usr/bin/myapp feature in voicemail.conf to chown the permissions to something else. Since they are root, asterisk should have no problem deleting and moving them around with less privileges. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: hugolivude [EMAIL PROTECTED] To: Asterisk-Users@lists.digium.com Sent: Tuesday, August 16, 2005 11:40 AM Subject: [Asterisk-Users] Voicemail file permissions I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). I'd like to give my Asterisk users the option of cleaning up their voicemail mailbox from their Windows PCs. I set up Samba and added all the users with restricted access to their mailbox only, but here's the problem: The voicemail .wav files that Asterisk creates have root as both owner and group. Since the users do not have root privileges, they can't do much with the files. BTW I'm not sure why the voicemail .wav files have root as both owner and group because I followed the instructions for running Asterisk other than root (see http://www.voip-info.org/wiki-Asterisk+non-root). Is there a way around this w/o giving everyone root privileges! Thanks, Hugh ___ 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] Polycom 501 dialing problem
Sounds like you have a DTMF mode problem. Check that you are using RFC2833 for dtmf signaling. I had the same thing happen with my dialing of *98 to check voicemail..It would transpose it in to 9*8, as if the * was being some sort of a tab key. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Craig Bruenderman [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, August 16, 2005 11:55 AM Subject: [Asterisk-Users] Polycom 501 dialing problem When I want to pick up a ringing line, I dial *8 and hit New Call softkey on my Poly 501. For some reason, if I pick up the hand set and dial *8, it seems to ignore or drop the 8 digit. I've confirmed that this happens with all of my 12 Polycom 501s. Does anyone know what would cause this or how to fix it? Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100 ___ 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] Transferring from cell phone
I set up a context to allow me to call in to my *
server (via Teliax in this case using IAX2) from my cellphone, and let me do a
number of things, including dial other extensions, AND dial outbound again so
callers could see my proper work CallerID when I use this service. Is
there a way to be able to transfer calls to other extensions of my asterisk
server FROM the cell phone/ This isn't a Zap channel, so I'm a bit lost,
but did specify the 'T' option in dial. Here's my context. Is this
possible to do??
[aa_chris_disa]exten =
s,1,Read(DIALNUM,custom/enter-num-then-pound,21)exten =
s,2,Playback(connecting)exten = s,3,GotoIf($[${LEN(${DIALNUM})} 5
]?4:8) ; IF SHORTED THAN 5, its internal so dial internalexten =
s,4,SetCallerID("Chris Mobile" 205)exten = s,5,Dial(Local/[EMAIL PROTECTED]/n)
;DIAL INTERNAL EXTENSIONexten = s,6,Playback(call-terminated)exten
= s,7,Goto(aa_chris_start,s,1)exten =
s,8,Gotoif($[${LEN(${DIALNUM})} = 7]?s,9:s,14) ;IF 7 DIGITS DIALED, ITS LOCAL,
PREPEND THE AREA CODEexten = s,9,SetCIDNum(99)exten =
s,10,Dial(${IPTRUNK}/360${DIALNUM},,T)exten = s,11,Dial(SIP/[EMAIL PROTECTED],,T)exten
= s,12,Playback(all-circuits-busy-now)exten =
s,13,Goto(aa_chris_start,s,1)exten = s,14,SetCIDNum(99) ;NUMBER
ISNT 7, OR LESS THAN 5 SO AREA CODE WAS ADDEDexten =
s,15,Dial(${IPTRUNK}/${DIALNUM},,T)exten = s,16,Dial(SIP/[EMAIL PROTECTED],,T)exten
= s,17,Playback(all-circuits-busy-now)exten =
s,18,Goto(aa_chris_start,s,1)exten =
i,1,Goto(aa_chris_start,s,1)
Chris Coulthurst
[EMAIL PROTECTED]
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[Asterisk-Users] Transfer a call from cell phone (pseudo-disa)
I set up a context to allow me to call in to my *
server (via Teliax in this case using IAX2) from my cellphone, and let me do a
number of things, including dial other extensions, AND dial outbound again so
callers could see my proper work CallerID when I use this service. Is
there a way to be able to transfer calls to other extensions of my asterisk
server FROM the cell phone/ This isn't a Zap channel, so I'm a bit lost,
but did specify the 'T' option in dial. Here's my context. Is this
possible to do??
[aa_chris_disa]exten =
s,1,Read(DIALNUM,custom/enter-num-then-pound,21)exten =
s,2,Playback(connecting)exten = s,3,GotoIf($[${LEN(${DIALNUM})} 5
]?4:8) ; IF SHORTED THAN 5, its internal so dial internalexten =
s,4,SetCallerID("Chris Mobile" 205)exten = s,5,Dial(Local/[EMAIL PROTECTED]/n)
;DIAL INTERNAL EXTENSIONexten = s,6,Playback(call-terminated)exten
= s,7,Goto(aa_chris_start,s,1)exten =
s,8,Gotoif($[${LEN(${DIALNUM})} = 7]?s,9:s,14) ;IF 7 DIGITS DIALED, ITS LOCAL,
PREPEND THE AREA CODEexten = s,9,SetCIDNum(99)exten =
s,10,Dial(${IPTRUNK}/360${DIALNUM},,T)exten = s,11,Dial(SIP/[EMAIL PROTECTED],,T)exten
= s,12,Playback(all-circuits-busy-now)exten =
s,13,Goto(aa_chris_start,s,1)exten = s,14,SetCIDNum(99) ;NUMBER
ISNT 7, OR LESS THAN 5 SO AREA CODE WAS ADDEDexten =
s,15,Dial(${IPTRUNK}/${DIALNUM},,T)exten = s,16,Dial(SIP/[EMAIL PROTECTED],,T)exten
= s,17,Playback(all-circuits-busy-now)exten =
s,18,Goto(aa_chris_start,s,1)exten =
i,1,Goto(aa_chris_start,s,1)
Chris Coulthurst
[EMAIL PROTECTED]
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[Asterisk-Users] Transfer a call from cell phone (pseudo-disa)
I set up a context to allow me to call in to my *
server (via Teliax in this case using IAX2) from my cellphone, and let me do a
number of things, including dial other extensions, AND dial outbound again so
callers could see my proper work CallerID when I use this service. Is
there a way to be able to transfer calls to other extensions of my asterisk
server FROM the cell phone/ This isn't a Zap channel, so I'm a bit lost,
but did specify the 'T' option in dial. Here's my context. Is this
possible to do??
[aa_chris_disa]exten =
s,1,Read(DIALNUM,custom/enter-num-then-pound,21)exten =
s,2,Playback(connecting)exten = s,3,GotoIf($[${LEN(${DIALNUM})} 5
]?4:8) ; IF SHORTED THAN 5, its internal so dial internalexten =
s,4,SetCallerID("Chris Mobile" 205)exten = s,5,Dial(Local/[EMAIL PROTECTED]/n)
;DIAL INTERNAL EXTENSIONexten = s,6,Playback(call-terminated)exten
= s,7,Goto(aa_chris_start,s,1)exten =
s,8,Gotoif($[${LEN(${DIALNUM})} = 7]?s,9:s,14) ;IF 7 DIGITS DIALED, ITS LOCAL,
PREPEND THE AREA CODEexten = s,9,SetCIDNum(99)exten =
s,10,Dial(${IPTRUNK}/360${DIALNUM},,T)exten = s,11,Dial(SIP/[EMAIL PROTECTED],,T)exten
= s,12,Playback(all-circuits-busy-now)exten =
s,13,Goto(aa_chris_start,s,1)exten = s,14,SetCIDNum(99) ;NUMBER
ISNT 7, OR LESS THAN 5 SO AREA CODE WAS ADDEDexten =
s,15,Dial(${IPTRUNK}/${DIALNUM},,T)exten = s,16,Dial(SIP/[EMAIL PROTECTED],,T)exten
= s,17,Playback(all-circuits-busy-now)exten =
s,18,Goto(aa_chris_start,s,1)exten =
i,1,Goto(aa_chris_start,s,1)
Chris Coulthurst
[EMAIL PROTECTED]
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Re: [Asterisk-Users] polycom 301 phone advice
I have two 300s and 4 500s. The 300s talk the same language, but have a lousy screen. The other thing to consider is, while it does have the 'monitor only' speaker, the volume is horrible. Cranked up to its highest setting, you can't hear voicemail with ANY background sound. Go for the 501 and sleep at night ;) Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Jim Duda [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, August 06, 2005 7:50 AM Subject: [Asterisk-Users] polycom 301 phone advice Can anyone tell me if the CallerID information is automatically displayed on the LCD screen of the 301? Can asterisk manipulate the LCD screen for the purposes of displaying callerid? Is this a good quality phone? Or, is the 501 worth the added expense? I believe the only real differences between 301 and 501 are that the 501 has one additional line (total of 3) and has speaker phone capability. Thanks, Jim ___ 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] I don't have option 5 in my voicemail
What do I put in voicemail.conf to let me send another user a voicemail from inside Comedian? I've CVS-HEAD, and the instructions are a bit ambiguous on the voicemaill.conf.sample. Advanced option 5 is the only on I don't have, and a very important one to have, indeed. Chris Coulthurst [EMAIL PROTECTED] ___ 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 SoundPoint 300 and distinctive ring
Yes it does apply. Near the top of your sip.cfg file, you should have lines
like this:
alertInfo voIpProt.SIP.alertInfo.1.value=ring-answer
voIpProt.SIP.alertInfo.1.class=4/
alertInfo voIpProt.SIP.alertInfo.2.value=internal
voIpProt.SIP.alertInfo.2.class=5/
alertInfo voIpProt.SIP.alertInfo.3.value=doorphone
voIpProt.SIP.alertInfo.3.class=6/
(I have a few here for auto-answer, internal extension ring cadence, and a
Zap doorphone alert)
You will also have something like these toward the bottom of sip.cfg under
ringType
RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer
se.rt.4.timeout=2000 se.rt.4.ringer=7/
INTERNAL se.rt.5.name=Internal se.rt.5.type=ring se.rt.5.ringer=3/
DOORPHONE se.rt.6.name=Doorphone se.rt.6.type=ring
se.rt.6.ringer=11/
Notice the connection between the class=4 on ring-answer above and below.
Duplicate these lines if you don't have them. Place them within the
SIP/SIP section and the ringType sections respectively. Pick your
ringer values based on the ones on the IP300 menu, which gives you chirps,
stutters, and trills etc.
Whatever value you have assigned (i.e. doorphone) is the value you must have
set in the _ALERT_INFO variable when you make the Dial(SIP) command:
[doorphone]
exten = s,1,Answer ;DOORPHONE IS CALLING
exten = s,2,SetCIDName(Doorphone 1)
exten = s,3,SetCIDNum(400)
exten = s,4,SetVar(_ALERT_INFO=doorphone) ;SET ALERT-INFO TO POLYCOMS
exten = s,5,Monitor(gsm,doorphone-${TIMESTAMP},m) ;RECORD THE DOORPHONE
CALL
exten = s,6,Dial(SIP/101SIP/102SIP/104SIP/201SIP/203Zap/2r3Zap/3,22)
;RING SOME PHONES
exten = s,7,Playback(nobody-but-chickens) ; NOBODY'S HOME
exten = s,8,Hangup
Note that you need the first underscore for ALERT_INFO if you are using
CVS-HEAD.
Hope that helps!
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: David Koski [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, August 04, 2005 10:00 PM
Subject: [Asterisk-Users] PolyCom SoundPoint 300 and distinctive ring
I am looking for clues on how to configure distinctive ring for a PolyCom
SoundPoint
300. Does ALERT_INFO apply? If so, how?
Thanks,
David Koski
[EMAIL PROTECTED]
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[Asterisk-Users] Voicemail advanced options, 5 to send a message not available
I never tried to use this feature until now, and noticed it isn't available to me according to the voice prompts. The documentation is a little vague on what is meant by 'context'..in voicemail.conf: sendvoicemail=yes ; Context to Send voicemail from [option 5 from the advanced menu] ; if not listed, sending messages from inside voicemail will not be ; permitted Why does this say "yes"? Is this referring to a context in extensions.conf, or is it a voicemail context (i.e. default)? Chris Coulthurst [EMAIL PROTECTED] ___ 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 and Presence
I have seen somewhat similar behavior, but I think its generally linked to me messing with the asterisk server. If, for example I restart asterisk and have voice mail waiting on a phone, it is right then that I will hear the reminder 'chirp' that the polycoms make to let you know its waiting. I think somethings are reset when I reload the chan_sip. Now this is just a guess, since I never seem to notice the presence state when I do this, or no one was actually on the phone. I have my line 2 and line 3 positions set up to show me two other sip extensions, so I don't have to press the 'buddies' button to see them. To reset it back to working, I just press 'directories' - Contact Directory - And edit the entry (without changing anything really), and pressing the SAVE button, and its all back again. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Polycom User [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, August 04, 2005 10:34 AM Subject: [Asterisk-Users] Polycom and Presence I am currently utilizing Polycom IP600 phones and presense. I have placed the hint directives in and everything seems to work fine. But this only works for a short period of time. After about 30 minutes, the extensions do not see when others are on the phone. Has anyone seen this type of behavior before? I am currently utilizing v1.5.2 of the polycom software and the cvs-head release from 8/3/05. Thanks in advance. ___ 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] Polycom phones w/ two lines on different servers
I can't offer you a config file since I no longer have a dual-server solution (I was just testing its feasibility) but I can tell you what I did. I didn't configure the second server on the sip.cnf files, I did it from the Browser page. The first server was an internal lan, which is also where the DNS comes from. The second server, I used the FQDN of the outside-world server. Other than usernames and passwords, I left all of the other fields as default. I have the IP 500 with 1.5.2.0054 SIP software and the OLDER 2.6.1 bootrom (for retro-compatibility). Oh, I also set 2 line appearances per CO key, rather than the default (I think it was 8?) Worked like a champ! Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, August 02, 2005 9:05 AM Subject: [Asterisk-Users] Polycom phones w/ two lines on different servers Hi all - This isn't really directly Asterisk related, but has anyone successfully set up a Polycom phone to register two lines on two different Asterisk boxes? I can get the first line to register, but the second one does not. I can still place calls from that second line, which indicates to me the server, user, and secret are correct. I'm running the newest 2.6 series firmware with the newest SIP image. My mac-sip.cfg contains: phone1 reg reg.1.displayName=1006 reg.1.address=[EMAIL PROTECTED] reg.1.label=1006 reg.1.type=private reg.1.auth.userId=1006 reg.1.auth.password= reg.1.server.dnsLookupOption=0 reg.1.server.1.address=192.168.0.90 reg.1.server.1.port=5060 reg.1.server.1.expires= reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.2.displayName=1003 reg.2.address=[EMAIL PROTECTED] reg.2.label=1003 reg.2.type=private reg.2.auth.userId=1003 reg.2.auth.password= reg.2.server.dnsLookupOption=0 reg.2.server.1.address=192.168.2.2 reg.2.server.1.port= The local overrides file for the phone contain: voIpProt.server.2.expires.lineSeize= voIpProt.server.2.retryMaxCount= voIpProt.server.2.retryTimeOut= voIpProt.server.2.register= voIpProt.server.2.expires=3600 voIpProt.server.2.transport=DNSnaptr voIpProt.server.2.port=5060 voIpProt.server.2.address=192.168.2.2 Any ideas? Sample configs would be appreciated if you've got something like this working in the past. Thanks, Nick ___ 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] DIDs in Thailand
Anyone know where to find a Thai DID to ring in SIP to asterisk? (probably Bangkok) Chris Coulthurst [EMAIL PROTECTED] ___ 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] Thailand DIDs
Anyone know where to find a Thai DID to ring in SIP to asterisk? (probably Bangkok) Chris Coulthurst [EMAIL PROTECTED] ___ 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] Thailand DIDs
Anyone know where to find a Thai DID to ring in SIP to asterisk? (probably Bangkok) Chris Coulthurst [EMAIL PROTECTED] ___ 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] Thailand DIDs
Anyone know of a place to get a Thailand DID that will ring in to asterisk in the US at a nice price? Chris Coulthurst [EMAIL PROTECTED] ___ 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] Thailand DIDs
Anyone know of a place to get a Thailand DID that will ring in to asterisk in the US at a nice price? Chris Coulthurst [EMAIL PROTECTED] ___ 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] Looking for Thai DIDs
Anybody know where to find Thailand DIDs that can ring in to my * in the USA on SIP? Oh, and a good price, too! ;) Chris Coulthurst [EMAIL PROTECTED] ___ 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
Apologies -- Re: [Asterisk-Users] Looking for Thai DIDs
My mail client left me the impression that it wasn't sending (software was locking up on SMTP password check). So sorry all... - Original Message - From: Chad Osmond [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, July 21, 2005 11:47 AM Subject: RE: [Asterisk-Users] Looking for Thai DIDs I do not know about Thailand DID's, but I would rather not see you post six times about this. There should be some information in the Wiki about providers all across the world and Google may have some additional information. Try the -biz list for biz' related questions. Or, the Wiki has a lot of information: http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers% 20by%20Country -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Coulthurst Sent: July 21, 2005 1:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Looking for Thai DIDs Anybody know where to find Thailand DIDs that can ring in to my * in the USA on SIP? Oh, and a good price, too! ;) ___ 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] COVAD voipr movie clip - A MUST SEE
If you haven't seen it yet, go here with a Flash enabled browser: http://www.theringingmovie.com Chris Coulthurst [EMAIL PROTECTED] ___ 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 phone digitmap question
As far as I know, the dialplan autodialer only works when the phone is off hook. This of course allows for nonstandard numbers to be dialed without regard to the digitmap. I, for example have lots of *XX numbers like *69 and *82, but if I wanted to dial *8 for a pickup I just dial *8 and then pickup the receiver. So, I guess in a way, its really a feature! ;) Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rudolf Ladyzhenskii |Sent: Friday, July 15, 2005 11:05 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] Polycom phone digitmap question | | |Hi, all | |I have Polycom SP300 phones. My extension range is 1xx, so I added |corresponding entry to the digitmap. | |By some reason this does not affect on-hook dialing. If I have phone |off-hook all is ok. dial extension 102 for example and it |connects. if phone is off-hooh, however, I have to press DIAL |or take it off hook |before number is sent. | |Any ideas? | |Thanks, |Rudolf |P.S. Happens on both SIP 1.3 and 1.5 firmware of SP300 | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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] How to 'read' ztmonitor and set gains
Being one the many people trying to track down echo 'ghosts' I ran across this page: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html suggesting ways to adjust the gain. I have a TDM400P 2x2 config with Kewlstart lines configured. I've located a local telco milliwatt test line, and when I call it, the gain numbers are no where near 14844. Now, this article refers to configuration with a channel bank, but suggests it would be similar on 'simpler equipment'. The numbers I get are around 4450 on the Rx. [EMAIL PROTECTED] zaptel]# ./ztmonitor 5 -vv Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) * Rx: 4452 ( 4452) Tx:46 ( 63) Is this good? Normal? Any suggestions, or a point in the right direction for the right documentation would be appreciated. P.S. This is CVS-HEAD Zaptel on a P3 550, Host bridge: Intel Corp. 440BX/ZX/DX - 82443BX/ZX/DX Host bridge (rev 3) Chris Coulthurst [EMAIL PROTECTED] ___ 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] Asterisk-CVS-HEAD locks up on 'reload' from CLI (sometimes)
Lately when I issue a 'reload' from the CLI, I find that it will sometimes hang forever, completely locked up. I can press enter and see the CLI prompt move, but no commands are taken. top shows asterisk eating everything up: PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 20669 root 25 0 10068 9.8M 5392 R88.4 1.9 1:02 0 asterisk 20877 root 15 0 1124 1124 896 R 0.3 0.2 0:00 0 top 1 root 15 0 448 448 396 S 0.0 0.0 0:04 0 init 2 root 15 0 00 0 SW0.0 0.0 0:01 0 keventd 3 root 15 0 00 0 SW0.0 0.0 0:00 0 kapmd 4 root 34 19 00 0 SWN 0.0 0.0 0:00 0 ksoftirqd/0 Most recent add-ons have been Speex and h323. I just installed h323 today and this has been going on for about a week, so I know its not that, but I can't remember if this was happening before Speex or not. Anyone have any similar happenings? Chris Coulthurst [EMAIL PROTECTED] ___ 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] Dial *97 to pickup voicemail buts says mypasswordincorrect
Not sure why I see *97 and *98 here, but I would check your dtmfmode= line in sip.conf. Often times, using rfc2833 works when inband or sip-info doesn't. See http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+dtmfmode Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Angus Comber |Sent: Monday, July 04, 2005 4:34 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Dial *97 to pickup voicemail |buts says mypasswordincorrect | | |I have found that if I dial from another extension *98 and |select extn 200 |and enter password 1234 it works. So is it something to do with |configuration on my IP Phone? It is a Grandstream GXP2000 running: |Software Version: Program-- 1.0.0.3Bootloader-- 1.0.0.3 | |Anyone got any ideas? | |Angus | | | |- Original Message - |From: Angus Comber |To: asterisk-users@lists.digium.com |Sent: Monday, July 04, 2005 12:20 PM |Subject: [Asterisk-Users] Dial *97 to pickup voicemail buts says my |passwordincorrect | | |Hello | |I am at extension 200 and I know there is a voicemail message |waiting. I |dial *97 and am prompted for the password. I enter 1234 which |I have set as |my voicemail password. What can I do to troubleshoot? | |Angus Comber |Itel Office Software Ltd |5 Enmore Gardens |London, SW14 8RF |Tel: 020 8878 7367 |Fax: 020 8876 7257 |Em: [EMAIL PROTECTED] |web: www.iteloffice.com | | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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/asteri|sk-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] Problems with zaptel and voice prompts/voicemail
I claim to be NO expert, but is there a chance that the 'ztdummy' driver is also being loaded? I'm thinking it might cause a timing conflict of some kind...I may be way off here, but I'd still check for something as simple as that... Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Jeremy McDermond |Sent: Wednesday, June 29, 2005 8:50 PM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Problems with zaptel and voice |prompts/voicemail | | |I've looked all around, and I can't find an answer to this. I |apologize if this has been discussed already or is buried somewhere |in voip-info.org. | |I have an asterisk setup on linux 2.6.11.11 kernel, a revision E/F |TDM400P, and Polycom IP501 phones. As soon as I load the zaptel |module into the kernel, the voice prompts and voicemail system ceases |to work. The asterisk logs say that the gsm files are being played, |but nothing comes out on the other end. This is for both calls |coming in via our VoicePulse Connect lines, or when dialing locally |from our SIP phones. As soon as I rmmod the zaptel driver, asterisk |acts just fine. | |Thanks for any assistance the list may be able to provide. |-- |Jeremy McDermond |Xenotropic Systems ___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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] Resolving groupcalls
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Martin Czarnowski
|Sent: Thursday, June 30, 2005 12:58 AM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Resolving groupcalls
|
|
|Hi,
|
|I'm trying to write a tool, which shows me the state of the current
|calls. For this purpose I'm reading from Pipe the Asterisk output and
|parse it... asterisk -vr | mytool
|
|However, the problem ist how to get the information about who got this
|call in the group. The Zap channels are assigned dynamical.
|Only thing I
|can see which channel is connect to the caller but not who is
|using the
|channel.
|
|I know there is the CDR output in Master.csv. But it shows me
|the same.
|The other problem with CDR is, that it shows me the Info only
|after the
|call is finished. That's why I'm trying to parse the asterisk output.
|
|My extensions.conf looks like this..
|GROUPCALL = Zap/g2/1200021Zap/g2/1200022Zap/g2/1200023
|.
|.
|exten = s,1,Dial(${GROUPCALL})
|
|
|___
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|http://lists.digium.com/mailman/listinfo/asteri|sk-users
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|UNSUBSCRIBE or update options visit:
| http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Resolving groupcalls
Oops, sent that last one prematurely!
How about the accountcode setting? You could get user information from
that, right?
Maybe you could send:
Asterisk -rx 'show channels'
..and when you get the data, you'd know which channels are up and alive
(full names).
You could then re-run the command with the channel information:
Asterisk -rx 'show channel SIP/201-ec69'
..you'd get a dump, with the end looking something like this:
CDR Variables:
level 1: clid=Chris Office 201
level 1: src=201
level 1: dst=18009427433
level 1: dcontext=unlimited
level 1: channel=SIP/201-ec69
level 1: dstchannel=IAX2/provider-7
level 1: lastapp=Dial
level 1: lastdata=iax2/[EMAIL PROTECTED]/2047622726
level 1: start=2005-06-30 02:10:35
level 1: answer=2005-06-30 02:10:38
level 1: end=2005-06-30 02:10:38
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: accountcode=019284718233 --account code
unique to the user
level 1: uniqueid=1120122635.400
Anyway, maybe something like that...
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Martin Czarnowski
|Sent: Thursday, June 30, 2005 12:58 AM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Resolving groupcalls
|
|
|Hi,
|
|I'm trying to write a tool, which shows me the state of the current
|calls. For this purpose I'm reading from Pipe the Asterisk output and
|parse it... asterisk -vr | mytool
|
|However, the problem ist how to get the information about who got this
|call in the group. The Zap channels are assigned dynamical.
|Only thing I
|can see which channel is connect to the caller but not who is
|using the
|channel.
|
|I know there is the CDR output in Master.csv. But it shows me
|the same.
|The other problem with CDR is, that it shows me the Info only
|after the
|call is finished. That's why I'm trying to parse the asterisk output.
|
|My extensions.conf looks like this..
|GROUPCALL = Zap/g2/1200021Zap/g2/1200022Zap/g2/1200023
|.
|.
|exten = s,1,Dial(${GROUPCALL})
|
|
|___
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|Asterisk-Users@lists.digium.com
|http://lists.digium.com/mailman/listinfo/asteri|sk-users
|To
|UNSUBSCRIBE or update options visit:
| http://lists.digium.com/mailman/listinfo/asterisk-users
|
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[Asterisk-Users] Recommend against Teliax as primary ITSP
I really hate to have to make a post like this, but I feel I have little choice but to relay to the group my experience with Teliax, and explain why I recommend against using them as a primary Voip- PSTN provider. I hope that a letter like this will inspire companies like Teliax to work harder at customer service, as well as circuit stability. We need more companies that offer the types of service they do. I have been using Teliax for about 3 months now, and they were my first ITSP when I started playing with Asterisk and my Grandstream BT101. I picked them arbitrarily because they had low rates, and supported the IAX2 protocol, which I determined to be more firewall friendly. Right away, I was happy with the reponse, the online ordering, and the low rate. It didn't take long for the multitude of outages to occur. Now, while I claim to be no VoIP expert, I did a variety of tests to make sure the problems weren't on my end. I recommended Teliax to a business partner, who has a Linux box in a data center downtown, and had access to their system as well. When I'd find outages, I would first check to see if they were having the same problem. So far, every time I had a problem, so did they. I am also registered with FWD on IAX2 and they were always up. Any tech support calls to Teliax would take more than 2 days to get a response. Only when I threatened to leave would someone suddenly pop up and answer my concerns. They claim to have been changing bandwidth providers (away from rockynet, or at least companies that peer with Cogent), but traceroutes show they are still with them. So far, when I've actually gotten ahold of a tech support person, they have told me to try different addresses for the server. They've changed recently from voip.teliax.com to ast01.teliax.com to voip-co1.teliax.com. Guess what? All the same server. Its just more of the same runaround. Since the day I switched to VoIP (with Teliax) as my primary outbound calling, more people have laughed at me for my choice of VoIP as a telco medium than can be counted. And these are people who respect me in the Telco community, and who I have been trying to convince of the benefits. They don't see the benefits when they can't call me, and I can't call them. I understand that all companies have their problems, especially with such emerging technology as VoIP. I would have very little problem with Teliax, and use a secondary provider as a backup, if they were more forthright in explaining their problems, and notify their customer base within a reasonable time when they are going to have outages due to network changes. As it stands, I now have to find a new provider that will at least duplicate the features of Teliax. The hardest part of this is, they offer BYOD, use IAX2, and let you change your callerid presentation. These are all things that I MUST have. If anyone has some positive results with a similar competitor, I'd love to hear about it. In the meantime, I have to change back over to PSTN lines temporarily, since I can't rely on service from Teliax. I hope any/all of you that use their service have better luck than I have with them. Chris Coulthurst [EMAIL PROTECTED] ___ 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] Teliax Problems
Try voip-co2.teliax.com to register with. And read my other letter I suppose. This domain is apparently working as of 4:30, but have had the same problem since 1:30 AM PDT. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Malcolm Taylor |Sent: Wednesday, June 29, 2005 4:14 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Teliax Problems | | | |I'm currently unable to register with Teliax's server via IAX2 |and can't reach them via either of their phone numbers. Their |website is up and I have logged a support incident. | |Is anyone else experiencing the same problems? Having been |caught up in the Broadvoice fiasco a couple of months back, |I'm hoping that Teliax is not going through the same sort of thing. | |Malcolm | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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] Teliax Problems
Does anyone have anything +/- to say about TeleSIP? They appear to have local DIDs where I live and all comments on the wiki indicate they are reputable.. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Wednesday, June 29, 2005 5:22 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Teliax Problems | | | I'm currently unable to register with Teliax's server via IAX2 and | can't reach them via either of their phone numbers. Their |website is | up and I have logged a support incident. | | Is anyone else experiencing the same problems? Having been |caught up | in the Broadvoice fiasco a couple of months back, I'm hoping that | Teliax is not going through the same sort of thing. | |An ethereal trace indicates the IP address is active, but it |is not responding to iax packets (registration). So, either |their asterisk app has failed or they have folded their tent as well. | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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] Teliax Problems
Yes I was just reading that TeleSIP and Telasip are often mistaken, and was just editing my dialplan for my mistakes! When you meen porting numbers, I assume you are talking about LNP? If so, not a problem for me anyway. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rick Baranowski |Sent: Wednesday, June 29, 2005 8:34 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Teliax Problems | | |I am assuming that you mean Telasip? | |Don't expect to get any numbers ported over to them. | |I have never been able to get anyone on the phone. | |Can't say that I have had any technical issues. | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Chris Coulthurst |Sent: Wednesday, June 29, 2005 4:49 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Teliax Problems | |Does anyone have anything +/- to say about TeleSIP? They |appear to have local DIDs where I live and all comments on the |wiki indicate they are reputable.. | |Chris Coulthurst |[EMAIL PROTECTED] | | | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of ||Rich Adamson ||Sent: Wednesday, June 29, 2005 5:22 AM ||To: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: Re: [Asterisk-Users] Teliax Problems || || || I'm currently unable to register with Teliax's server via IAX2 and || can't reach them via either of their phone numbers. Their ||website is || up and I have logged a support incident. || || Is anyone else experiencing the same problems? Having been ||caught up || in the Broadvoice fiasco a couple of months back, I'm hoping that || Teliax is not going through the same sort of thing. || ||An ethereal trace indicates the IP address is active, but it ||is not responding to iax packets (registration). So, either ||their asterisk app has failed or they have folded their tent as well. || || ||___ ||Asterisk-Users mailing list ||Asterisk-Users@lists.digium.com ||http://lists.digium.com/mailman/listinfo/asteri|sk-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 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] Polycom SoundPoint 501 Problem
Title: Message Need some more information about your setup. If you go to the web interface (IP of your phone on browser), and check out the "Lines" section, do you have the correct Auth User ID, Address and Password set? In the same section, make sure that you are using the right SIP port of (usually) 5060 and the right address, as well as having a '1' in the REGISTER box. See if that is all correct. Without more config information, its just a shot in the dark. Chris Coulthurst [EMAIL PROTECTED] -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig BruendermanSent: Wednesday, June 29, 2005 11:49 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Polycom SoundPoint 501 Problem I'm attempting to set up my SoundPoint 501 with my Asterisk server. I've configured DHCP and TFTP and successfully updated both the BootRom and SIP application. I've also created a custom cfg file for this phone's MAC address and the settings seem to be taking just fine. I can see that the phone registers with my Asterisk server but 'sip show peers' reports that the phone is UNREACHABLE. This seems to be a problem because when I dial an extension from the phone, I can see Asterisk answering it and Festival responding, but I get no audio out of the handset nor the speaker of the phone. I'm running Asterisk stable and the phones are 2.6.2 bootrom with SIP 1.5.2. Any clues? Craig Bruenderman Network Advocates, Inc.9001 Shelbyville Road, 260 BurhansLouisville, KY 40222Main: 502-412-1050Free: 877-412-1050DID: 502-992-5929Fax: 502-412-1058Mobile: 502-548-1100 ___ 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] APP - ValetParking on CVS-HEAD -- instructions on its use, anyone?
I've finally got this beast installed, but now I don't see how to use it. I've looked over the web near and far, and it just seems that no one has anything on its implementation.. This is all I get from the CLI: *CLI -= Info about application 'ValetParking' =- [Synopsis] Valet Parking [Description] ValetParking(exten|lotname|timeout[|return_ext][|return_pri][|ret urn_context]) Auto-Sense Valet Parking: if exten is not occupied, park it, if it is already parked, bridge to it. I can guess that everything after 'timeout' is optional because of the brackets, but I'm confused on everything else. What is the 'exten'? I was under the impression that it was auto-generating on this particular app. I have no idea what 'lotname' is, but I feel that there should maybe be some lines added to features.conf for this thing? Just no docs to tell me what to do next. Help, anybody? I'd love some real-dialplan working examples... Desparately yours, Chris Coulthurst [EMAIL PROTECTED] ___ 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 and CallerID
Which software pack to you have for the IP600? Sip.ld, bootrom, etc... Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of Johann |Sent: Tuesday, June 21, 2005 11:54 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] Polycom and CallerID | | |I'm having a problem with the callerID that the polycom IP600 |phones are |displaying. I would like to modify the CIDName and leave CIDNumber as |exactly what the phone call came in as(provided they aren't hiding |callerID). Most of the calls will be going to the queue, but |a few will |go directly to the SIP phones. | |I've done a various combinations of using SetCallerID(), SetCIDName(), |and SetCIDNum(). It seems the most I can get is to change the CIDName |that the polycom phones will show. The CIDNumber is always simply |asterisk. I have put NoOp statements before the call hits the queue |and the dial() command to ensure that it being set right. It shows up |in the Asterisk CLI correctly. The SetCallerID() does set the name, |however SetCIDName() and SetCIDNum() do not work. The contact |is always |listed as asterisk. | |Anyone else have any problems with callerID, asterisk, and |polycom(IP600) phones? | |--johann |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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] 2nd Dialtone after answer
Check out DISA. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Oswaldo Arratia |Sent: Friday, June 17, 2005 7:51 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] 2nd Dialtone after answer | | |Hi |I am trying to achive this for a specific need of a customer. | |He has a DID pointed to an Asterisk server, I need to provide |him dialtone when the calls hits the server. How can I achieve this? | |Let's say something like this: | |Exten = s,1,Answer |Exten = s,2, Provide Dial tone |Exten = s,3, Dial the number the person will enter after |receiving the dial tone Exten = s,4,Hangup | |Any ideas? | |Thanks very much | |Oswaldo | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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] DID Issue
If you have a loopback plug, I would take that PRI down, unplug the NIU from the Asterisk box, and plug that RJ45 loopback plug in to the NIU, and call the telco, have them run a loop test on your circuit. Out here in Qwest-land they can usually get a tester on it and get results to you in less than an hour. Sounds to me like that problem is theirs, this would help prove it. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Mark Johnson |Sent: Sunday, June 12, 2005 7:53 AM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] DID Issue | | |I have a pretty strange problem. I have about 100 DID's that |come down |a PRI from SBC in the United States. On Friday afternoon, one of my |DID's flipped out. When you call it, the SBC operator comes |on and says |that the line has been disconnected. I contacted them and |they ran test |and they are telling me the problem has to be on my end. My |problem is |that the CLI never shows the number as called. It seems to me |it would |show that ZAP channel ring and then say what it decided to with it. |I've got nothing. I even shut the * box down and brought it back up, |same problem... | |In the past, if I shut down a SIP device and you try to call the DID, |I'm pretty sure you got a busy signal, not an SBC operator. |Anyone have |idea how to troubleshoot this one? I pretty sure it's a problem with |the phone company, some type of translation issue. | |Mark |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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] DID Issue
At some time with Qwest for my Frame Relay and 24 voice lines, during all my similar long-running headaches, they inadvertently gave me a 'direct' tier 2 tech number. Now normally, you'd expect a trouble ticket # to be handed to these guys, but they never asked for more than my Circuit ID # or the pilot # of the DIDs, and helped me right from the switchroom! If you are lucky enough to get a number like this, its GOLD. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Mark Johnson |Sent: Sunday, June 12, 2005 1:31 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] DID Issue | | |Chris Coulthurst wrote: | |If you have a loopback plug, I would take that PRI down, |unplug the NIU |from the Asterisk box, and plug that RJ45 loopback plug in to |the NIU, |and call the telco, have them run a loop test on your circuit. Out |here in Qwest-land they can usually get a tester on it and |get results |to you in less than an hour. Sounds to me like that problem |is theirs, |this would help prove it. | |Chris Coulthurst |[EMAIL PROTECTED] | | | |After arguing with them for the last few days, they finally discovered |the problem was on their end. They somehow lost the DID in the |translation database. They simply added it back and it works. What |upsets me is that they insisted my equipment was telling them |it was an |unlocated number. It's tough to argue with a large phone company that |they are wrong!! | |Mark |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-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] how to tell
Title: Message With DEBUG set to 50 (probably much less would do it) I get this message: 2005-06-12 14:06:06 DEBUG[31276]: Got event Wink/Flash(3) on channel 2 (index 0)2005-06-12 14:06:06 DEBUG[31276]: Winkflash, index: 0, normal: 22, callwait: -1, thirdcall: -12005-06-12 14:06:06 DEBUG[31276]: Already have a dsp on Zap/2-2?2005-06-12 14:06:06 DEBUG[31276]: Swapping 2 and 02005-06-12 14:06:06 DEBUG[31276]: disabled echo cancellation on channel 22005-06-12 14:06:06 VERBOSE[31280]: -- Starting simple switch on 'Zap/2-2'2005-06-12 14:06:06 VERBOSE[31276]: -- Started three way call on channel 2 Is that what you wanted? Chris Coulthurst [EMAIL PROTECTED] -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd A. RikerSent: Sunday, June 12, 2005 1:41 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] how to tell How do I tell if Asterisk is receiving a flash hook command? 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
RE: [Asterisk-Users] how to tell
Title: Message Reading a little bit on the web, Im getting the impression this may not be easy to do with the SPA-2002. Apparently Asterisk expects an RTP "DTMF named event" 16 to interpret a FXS flash over a SIP line. I may be completely wrong about this but I was reading this from this page refering to a Linksys device with similar problems. Also mentioned on another was that the SPA-3000 solves these flash issues. Read these if you'd like: http://bugs.digium.com/view.php?id=4283nbn=8 http://voxilla.com/modules.php?op=modloadname=Newsfile=articlesid=161 Hope any of that helps... Chris Coulthurst [EMAIL PROTECTED] -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd A. RikerSent: Sunday, June 12, 2005 2:17 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] how to tell This helps, Chris. When I press the flash key on my analog phone (connected to SPA-2002), Asterisk does not display this info. I believe that Asterisk is either not receiving the hook flash from the Sipura or the Sipura is intercepting it and not passing it to the Asterisk box. I have call waiting on my PSTN line, and I cannot flash the X100P card to switch to the other call. Do you have any ideas? Thanks From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris CoulthurstSent: Sunday, June 12, 2005 4:09 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] how to tell With DEBUG set to 50 (probably much less would do it) I get this message: 2005-06-12 14:06:06 DEBUG[31276]: Got event Wink/Flash(3) on channel 2 (index 0)2005-06-12 14:06:06 DEBUG[31276]: Winkflash, index: 0, normal: 22, callwait: -1, thirdcall: -12005-06-12 14:06:06 DEBUG[31276]: Already have a dsp on Zap/2-2?2005-06-12 14:06:06 DEBUG[31276]: Swapping 2 and 02005-06-12 14:06:06 DEBUG[31276]: disabled echo cancellation on channel 22005-06-12 14:06:06 VERBOSE[31280]: -- Starting simple switch on 'Zap/2-2'2005-06-12 14:06:06 VERBOSE[31276]: -- Started three way call on channel 2 Is that what you wanted? Chris Coulthurst [EMAIL PROTECTED] -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd A. RikerSent: Sunday, June 12, 2005 1:41 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] how to tell How do I tell if Asterisk is receiving a flash hook command? 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
[Asterisk-Users] *66 auto redial emulation?
Has anyone ever tried to roll out a *66 auto-callback-redial feature on
asterisk?
I'm sure that implementing this for outbound Zap calls would be a nightmare,
but what about something easier, like internal extensions?
On my old Panasonic key system, it used to be such that, if the called
extensions were busy, you could press 6 while hearing the busy signal, it
would beep twice and hangup. Once the called, busy party's line was free,
the KSU would ring-ring your phone back, and once you picked up, it redialed
the extension that is freed.
I have a rough idea on how to implement something like this, but I can see a
few complex issues.
1. IF you want it to ring busy, and have the same behaviors as the Panasonic
system did, you can't TRULY ring busy. Need a playtones command here to go
to instead. I have already sketched one out but it needs work.
2. If the called party actually has call waiting, maybe we could still get
the 'busy' tone but instead of pressing 6 to callback, we press 1 to signal
their call-waiting second line. If that line is also busy (they have a
caller on hold), then we could maybe go to their VM at this point.
Okay, are we confused enough yet? You bet, I sure am, and would love any
input and sample extensions.conf context examples you may have. Here is the
one that I kinda-wrote out for the busy-generation:
[new-busy]
exten = s,1,Answer()
exten = s,2,ResponseTimeout(60)
exten = s,3,Playtones(busy)
exten = s,4,Read(dialed,,1)
exten = s,5,Goto(${dialed},1)
exten = 1,1,StopPlayTones
exten = 1,2,** GO DO SOMETHING WHEN YOU PRESS 1 HERE **
exten = 1,3,Hangup
exten = 6,1,StopPlayTones
exten = 6,2,** GO DO SOMETHING ELSE WHEN YOU PRESS 2 **
exten = 6,3,Hangup
exten = i,1,Goto(s,3); everything else is invalid
exten = t,1,Hangup ; busy timed out after 60 seconds
Chris Coulthurst
[EMAIL PROTECTED]
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RE: [Asterisk-Users] Newie Questions RE: Polycom Critique
I have 4 Polycom phones here, two 500s and two 300s. The 500 is a top-shelf phone with quite a few asterisk-friendly features. I absolutely love the speakerphone: it has superb tone quality, and its truly full-duplex. The caller on the speaker does NOT hear him/herself back through the microphone. I can only assume some level of active noise reduction. The 300 Model has no speakerphone, just a monitor-mode speaker, and the sound quality isn't as good, but its still very competitive... If you choose a Polycom, you are choosing 'programmability'. While quite simple to provision out-of-the-box to just get going, if you want to fine-tune feature keys, alert-info types and the like, you will be digging through the rather thick admin guide at first, but soon you start remembering where to look for a change. Asterisk apparently does not fully/partially support the SIP SUBSCRIBE messages this phone wants to use for CallPark, GroupPickup, etc. Once this becomes possible, I doubt there is much you can't duplicate like a traditional key-system. The only gripes I really have about this phone are these: There is NO BACKLIGHT. C'mon companies! My old Panasonic KX-TD1232 is 12 years old, and has no backlight. I'm ready for this little bit of sunshine! It really surprises me, when I look at a entry-level low-budget phone like the Grandstream BT100 and see that even it has a basic blue backlight. The other gripe I have is, Polycom doesn't well-document any of the 'enhanced' features showcased on these phones. There is a SERVICES button that seems to have no purpose on the 500, but brings up a minibrowser on the 600. Presence and SIMPLE aren't well documented/possible with asterisk yet, and what the Polycoms do offer is extremely limited in its practicality. All in all, if Polycom would put in a little backlight, and make a matching SIP-enabled DSS console with REAL LEDs, I'd run with them and never look back! Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor |Sent: Friday, June 10, 2005 9:29 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Newie Questions | |Thanks for your repsonse, perhaps I mis-stated my situation. I have |asterisk up and running with a TDM22B and have two analog phones working |with two analog phone lines. What I can't seem to get started on is the |setup of a SIP phone. I have looked at all the info on voip-info.org |and it is somewhat helpful, but not enough to get it going. So any help |would be appreciated. | |Also, is it generally accecpted that the Polycom phones are a good |choice? Why might I choose something else? Can the Polycom phones be |setup to work against a propritary phone system like the Nortel or Avaya? | |Thanks again, | |Matt | | | |Dean Collins wrote: | |Yes asterisk not only competes with avaya and Nortel but exceeds them once |you know what you are doing. | |If you are only new to Asterisk there is now [EMAIL PROTECTED] |http://asteriskathome.sourceforge.net | |don't be put off by the name - people run entire companies on this |version) |The [EMAIL PROTECTED] solution the easiest way to get started. It is an .iso |cd that you burn, load into a suitable PC (I run mine on a P3-700) and this |super smart scripting code automatically installs the following software; |Asterisk (the open source switching software) |AMP (an open source release of a gui configurator) they have their own |separate sourceforge website https://sourceforge.net/projects/amportal |FOP (a graphical web page for transferring calls, monitoring who is online |etc) http://www.asternic.org |Web meetme (a graphical web page for monitoring and controlling conference |calls) | |Check out www.voip-info.org for information about configuring your Polycom | |Welcome to the family. | |Cheers, |Dean | | | | |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor |Sent: Friday, 10 June 2005 5:27 PM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Newie Questions | |Hello, I'm new to asterisk. My company is opening a new office and I'm |seriously considering using Asterisk for the phone system. | |A couple of questions: | |How does Asterisk compete with the Avaya IP Office or the Nortel BCM |systems? | |I have purchased a Polycom 500 phone but I'm having trouble getting it |setup and talking to Asterisk. Is there somewhere that has SIP phone |setup A-Z for beginners? All the documentation I have seen assumes you |know more than I know at this point. | |I'm sure I'll have lots more questions, but that will do for now. | |Thanks, | |Matt | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http
RE: [Asterisk-Users] VOIP-INFO
It sounds like there are quite a few people willing to aid in bandwidth for voip-info. I was just wondering if it wouldn't make sense to mirror the site across several locations with a round-robin DNS for a little bit of load balancing? Any thoughts? Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Nir Simionovich |Sent: Friday, June 10, 2005 6:03 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] VOIP-INFO | |If required, I'd be more than happy and willing to let voip-info.org be |hosted on my hosting server. |We are currently hooked up to the net with a 6MB symetrical connection, |and it should be enough |for voip-info. In addition, I can perform a daily incremental back to |it, in the same manner I backup |all the other hosted site. | |voip-info is one of the most valuable tools around, and having it go |down on us is a disaster to everybody. | |Nir S | |Andrew Kohlsmith wrote: | |On Friday 10 June 2005 02:28, Olle E. Johansson wrote: | | |I would like to use this moment to say a big THANK YOU from the |community to you and Commpartners for providing this resource to the |community... | | | |I agree; while I personally dislike wikis I can't deny (as is evidenced by |all |the posts here in this thread) that voip-info.org is a very important |resource for this community, and I'm sure that it is a mostly thankless |job |to boot. | |Thank you, James, for the blood sweat and tears, not to mention money, |that |you pour into voip-info.org. | |-A. |___ |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 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] Convert extensions.conf INTO MySQL script
I swear I read somewhere on one of the MANY pages that there is a script out there that can read the extensions.conf file and create the MySQL DB records on the fly. Anyone know where I look for such a thing? Sure speeds up migration! Chris Coulthurst [EMAIL PROTECTED] ___ 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] REPOSTED: Polycom 500 Group Call Pickup Feature and *
If you activate (via sip.cfg) the feature Group Call Pickup, its no surprise that asterisk doesn't know what to do with this feature request. But it is being sent as a SIP SUBSCRIBE request, and I'm wondering if, as asterisk stands, there is a way to take advantage of this feature to emulate the *8# normal behavior. If anyone has any input, there is also a call parking function that I think is SIP SUBSCRIBE-based. Here is the 'sip debug' snippet from when I pressed the New Call - Pickup - Group softkeys: Sip read: SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bKa58a6cc24AEA0129 From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED] CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Accept: application/dialog-info+xml Max-Forwards: 70 Expires: 0 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.168.0.234 : 5060 (non-NAT) Found peer '201' Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bKa58a6cc24AEA0129 From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED];tag=as1b873db6 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=5041eff0 Content-Length: 0 to 192.168.0.234:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms morse*CLI Sip read: SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bK802f53579213D6EA From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED] CSeq: 2 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Accept: application/dialog-info+xml Proxy-Authorization: Digest username=201, realm=asterisk, nonce=5041eff0, uri=sip:[EMAIL PROTECTED]:5060, response=b48b989d85958a6ce18c9431058ce6f3, algorithm=MD5 Max-Forwards: 70 Expires: 0 Content-Length: 0 15 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.168.0.234 : 5060 (non-NAT) Found peer '201' Looking for groupcallpickup in default Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bK802f53579213D6EA From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED];tag=as1b873db6 Call-ID: [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.234:5060 Destroying call '[EMAIL PROTECTED]' morse*CLI sip no debug SIP Debugging Disabled Chris Coulthurst [EMAIL PROTECTED] ___ 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] voicemail check for busy message
How about the System()
command? I havent tried it
but I wonder if something like this would work
Exten = 1234,1,System(/usr/bin/head
/var/spool/asterisk/vm/default/${EXTEN}/INBOX/busy.gsm)
Exten = 1234,2,Voicemail(u${EXTEN})
Exten = 1234,102,Voicemail(b${EXTEN})
As I understand it, System()
will return -1 on errors (such as FILE NOT FOUND) and continue to priority
n+101. Im not sure if that
means -1 if the program spits out an error, or if
RUNNING the program itself is unsuccessful. Might be that it returns 0 even if
the file doesnt exist, because as in my example, the program /usr/bin/head DOES exist.
Hey, if it works, post it!
Chris Coulthurst
[EMAIL PROTECTED]
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Sturtevant
Sent: Thursday, June 09, 2005 1:25
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
voicemail check for busy message
Is there a way to check to see if a user has recorded a busy
message? If they havent I would prefer to send them to
u and play their unavailable greeting. I know I can
send them to u in both cases, but I would like to send them to
b if a recorded busy greeting exists.
Thanks!
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[Asterisk-Users] VOIP-INFO
Anyone else unable to get to www.voip-info.org? Site is returning 'connection refused' here. Chris Coulthurst [EMAIL PROTECTED] ___ 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] VOIP-INFO
You know, if I thought that it wouldn't be a bandwidth hog, I'd happily host voip-info out of my Data Center leased cabinet, but whoa-nellie! I think we know how popular that site is, hence why it can be a slow-loader at times. It would cost me a $FORTUNE$ with the price I pay per MB! I wonder if voip-info would let anyone 'snake' the site for backup? Who is the authority for that site? Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Neal Walton |Sent: Thursday, June 09, 2005 3:17 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] VOIP-INFO | | |I queried the name servers and it looks like this site may be hosted on a |cable or DSL system which doesn't allow static IP addresses. It's possible |that the DHCP and the dynamic DNS fell out of sync. All of this is just a |guess based on the name server results. | |On Thursday, June 09, 2005 2:49 PM, Wiley Siler |[SMTP:[EMAIL PROTECTED] wrote: | Been that way a couple of hours I think | W | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] On Behalf Of Chris | Coulthurst | Sent: Thursday, June 09, 2005 2:34 PM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: [Asterisk-Users] VOIP-INFO | | Anyone else unable to get to www.voip-info.org? Site is returning | 'connection refused' here. | | Chris Coulthurst | [EMAIL PROTECTED] | | | | ___ | 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 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] VOIP-INFO
Yeah funny you just heard that because I came on here now to post that the Google-cached front page of voip-info.org apparently said this: VOIP-info.org will be down for a few hours on Thursday June 9 for a system upgrade. I mean, who uses the front door of a website anymore? ;) I eagerly look forward to the rebirth... Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Peter A. Solomon |Sent: Thursday, June 09, 2005 9:20 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] VOIP-INFO | |Good News, according to Jim at Voip-info.org, they are doing an upgrade or |something to the server tonight at it will be back on line shortly. They |are |located in Hawaii and AHST. | |Pete | |___ |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] Voicemail and MS Exchange Synchronization
Just to put my two cents in..
Whatever the solution for integration with Outlook, I don't think any of the
voicemail should be downloaded via IMAP, POP or any other mail protocol. I
also think that it should be possible to playback the voicemail in outlook
with a suitable plugin. So, it would seem that a method be implemented to
allow an authenticated user to check them directly from the .wav-stored
location, or the like, so you could fast forward, rewind, etc.
Where I like the idea of using the telephone for the voicemail at all is, if
you decide to return the call or leave a message for someone else's mailbox,
a TAPI client could initiate the station phone to ring, user picks it up and
the call goes out the door.
ShoreTel does something similar to this with their system; while the
voicemail appears to be downloaded by Outlook, it is merely being displayed
within the Outlook GUI, and is really only retrieved when selected (like
IMAP does, but without the IMAP part).
Seems to me that writing macros, plugins, etc. for Outlook is rather easy
form Visual Basic. My brother is an extremely capable VB programmer, and I
think if the asterisk community were to adopt a more-or-less standard way of
binding to his Outlook app, we could make something that worked pretty good
and be portable to other programs other than Outlook.
I'm thinking of an API on the asterisk server that would complement the best
features of the manager api, the CLI and direct shell access to voicemail
directories and the like.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Kris Boutilier
|Sent: Thursday, June 09, 2005 9:40 PM
|To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
|Discussion
|Subject: RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization
|Importance: Low
|
| -Original Message-
| From: [EMAIL PROTECTED]
| [mailto:[EMAIL PROTECTED] Behalf Of Tim
| Litwiller
| Sent: Thursday, June 09, 2005 9:16 PM
| To: Asterisk Users Mailing List - Non-Commercial Discussion
| Subject: Re: [Asterisk-Users] Voicemail and MS Exchange
| Synchronization
|
|
| I'm not a programmer - but it sounds to me like you are all
| making it to hard by transfer the voice files around etc. unless you
| really have to have the messages stored in the mail server for some
|reason.
|
| here is what I would picture
| a outlook plugin that creates the illusion of several folders
|{clip}
|
|This is good, but just the configuration management process quickly becomes
|unrealistic for a larger office deployment of, say, 250+ clients.
|Similarly, a larger organisaion running Exchange presumably has also
|architected their hardware for it and would get better value from
|consolidating storage in that hardware - consider someone running an
|Exchange Server cluster with a SAN behind it...
|
|Perhaps there really is a need for two tiers of solutions here - the large
|scale Mailserver-as-backend-for-Asterisk concept presented earlier today by
|Craig Guy and then a more generic Asterisk-as-backend-for-mail clients
|using IMAP, some other mail protocol or even a per-mail client custom plug-
|in as you've suggested.
|
|Just my can$0.02
|
|Kris Boutilier
|Information Services Coordinator
|Sunshine Coast Regional District
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[Asterisk-Users] Polycom 500 Group Call Pickup Feature and *
If you activate (via sip.cfg) the feature Group Call Pickup, its no surprise that asterisk doesn't know what to do with this feature request. But it is being sent as a SIP SUBSCRIBE request, and I'm wondering if, as asterisk stands, there is a way to take advantage of this feature to emulate the *8# normal behavior. If anyone has any input, there is also a call parking function that I think is SIP SUBSCRIBE-based. Here is the 'sip debug' snippet from when I pressed the New Call - Pickup - Group softkeys: Sip read: SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bKa58a6cc24AEA0129 From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED] CSeq: 1 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Accept: application/dialog-info+xml Max-Forwards: 70 Expires: 0 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.168.0.234 : 5060 (non-NAT) Found peer '201' Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bKa58a6cc24AEA0129 From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED];tag=as1b873db6 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=5041eff0 Content-Length: 0 to 192.168.0.234:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms morse*CLI Sip read: SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bK802f53579213D6EA From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED] CSeq: 2 SUBSCRIBE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Accept: application/dialog-info+xml Proxy-Authorization: Digest username=201, realm=asterisk, nonce=5041eff0, uri=sip:[EMAIL PROTECTED]:5060, response=b48b989d85958a6ce18c9431058ce6f3, algorithm=MD5 Max-Forwards: 70 Expires: 0 Content-Length: 0 15 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.168.0.234 : 5060 (non-NAT) Found peer '201' Looking for groupcallpickup in default Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.234;branch=z9hG4bK802f53579213D6EA From: Chris Office sip:[EMAIL PROTECTED];tag=569A308-31C12E4D To: sip:[EMAIL PROTECTED];tag=as1b873db6 Call-ID: [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.234:5060 Destroying call '[EMAIL PROTECTED]' morse*CLI sip no debug SIP Debugging Disabled Chris Coulthurst [EMAIL PROTECTED] ___ 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] 3com 3105 Attendant DSS Console (SIP??)
Browsing, I came across this 3com DSS console: http://www.3com.com/products/en_US/detail.jsp?tab=featurespathtype=purc hasesku=3C10405A The ad claims SIP by Summer 2005. Does anyone know anything about this device and its interoperability? Any potential asterisk support/integration with something like this? Chris Coulthurst [EMAIL PROTECTED] ___ 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] run a script on completion of call
http://www.voip-info.org/wiki-Asterisk+h+extension This might help Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Eric Smith - Fruitcom |Sent: Tuesday, June 07, 2005 1:02 AM |To: Asterisk Users Mailing List |Subject: [Asterisk-Users] run a script on completion of call | |How do I run an external script on completing a call? | |Like if I want to send email to the caller. | |Thansk | |Eric |___ |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] Polycom 500 'SERVICE'S' key
Does anyone know if there is any way to make the 'services' key do anything? How about a way to remap it? Chris Coulthurst [EMAIL PROTECTED] ___ 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 Phones shorter than /24 netmasks
I just assigned one yesterday in a 10.X.X.X network with a netmast of 255.0.0.0, and had no problems..FYI. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Charlie Watts |Sent: Tuesday, June 07, 2005 9:25 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] Polycom Phones shorter than /24 netmasks | |Has anybody tried to use a Polycom phone (I have 500s and 600s) with a |netmask shorter than /24? (A network bigger than 255.255.255.0). We've |run out of IPs in our initial /24 network, and I'd like to expand it to |255.255.248.0. | |When I set it to 255.255.248.0 I can ping the phone while the bootloader |has control. As soon as the SIP application starts, I stop getting ping |responses. Phone never registers. | |255.255.254.0 seems to work, 255.255.252.0 doesn't. | | |My printers, servers, desktops, switches ... Everything else is fine. | |Comments? | |-- |Charlie Watts |[EMAIL PROTECTED] |Mercury Payment Systems - Information Technology |970-385-3187, 800-846-4472 x 3187 | | |___ |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] Polycom 500 'SERVICE'S' key
The manual implies that the services button (#31) is not among the programmable buttons. Have you had success with this key at all? The only reference to the services button in the manual beyond the keymap is that the Polycom 600 can use the minibrowser functions when this key is selected. Hard to imagine that Polycom would add this button on the 500, an obviously different phone layout, unless they plan to add it as a feature later. I wish I knew how to program my own sip.ld files. I could make this phone 'asterisk-specific'!! :) Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Charlie Watts |Sent: Tuesday, June 07, 2005 9:04 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Polycom 500 'SERVICE'S' key | |Chris Coulthurst wrote: | Does anyone know if there is any way to make the 'services' key do | anything? | | How about a way to remap it? | |Look in the Polycom SIP Administrators guide. You can map the button to |a variety of other things. | |I haven't tried it, but the documentation appears fairly clear. |___ |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] SetCallerID based on extension
If you're not using the ${ACCOUNTCODE} variable, you could do what I
did:
Here is an example of a sip.conf entry I have, stripped of non-relevant
detail:
[109]
accountcode=206555109 notice this is a combo of cid for trunks
-and extensions fused together
context=default
type=friend
username=109
host=dynamic
allow=ulaw
callerid=Chris Office 201
Now when anyone calls intra-office, their 'callerid' extension shows up.
Here's some of my extensions.conf:
[trunk-outbound]
exten = _1NXXNXX,1,SetCIDNum(${ACCOUNTCODE:0:10})
exten = _1NXXNXX,2,Dial(${OUT-TRUNK}/${EXTEN})
exten = _1NXXNXX,3,Congestion()
exten = _1NXXNXX,103,Busy()
The SetCIDNum is all I need for Caller ID from Teliax, and the :0:10
strips the first 10 digits off the ACCOUNTCODE to make it work.
I'm sure there are prettier ways, but its something to consider using if
you have no need for account codes.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Jodie Crouch
|Sent: Saturday, June 04, 2005 10:39 PM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] SetCallerID based on extension
|
|I am going through previous posts, but I am not finding anything. I
|apologize if this has been covered already.
|
|I want to be able to change the CallerID for outbound calls based on
|the extension making the calls. However, I don't really need this
|level of granularity. I have one asterisk box serving many markets.
|I want to set the CallerID to a local number in each market. So, of
|someone in Dallas, TX calls out it shows our Dallas number. If
|someone in Arizona calls out, I want it to show our Arizona number. I
|figured the best way to do this would be based on the extension, but
|if someone else has a better idea, please let me know.
|
|So, if we are going to set the caller id based on the extension, then
|this is what I need to know how to do.
|
|Right now, outbound is configured as below, in the extensions.conf
file.
|
|exten = _1NX,1,Dial(SIP/9725432876/${EXTEN})
|exten = _1NX,2,Hangup
|
|
|Please let me know how I can create this same process but change the
|CallerID to 2134531762 instead of 9725432876 when someone calls from
|extension 112 and to 9725432876 for extension 317.
|
|Thanks!
|Jodie
|___
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[Asterisk-Users] Automatic callback feature *66
Does anyone have a quick-n-dirty context to implement *66 automatic callbacks? I have a few people who like to have no call waiting on their phone (can you really blame them?) It would be nice to have something like *66, and also like 'Camp On', but instead of waiting something like 30 seconds, monitor the channel until it becomes available, then immediately ring back your phone to initiate the call. Ok, I redundantly described what it does. Anyone have a way to make it happen? Thanks much... Chris Coulthurst [EMAIL PROTECTED] ___ 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] Caller ID Routing using VoicePulseConnect
I would check the log when an inbound call comes in and see how VoicePulse is sending you the caller's digits. I use Teliax, and they send a +12035551212 format. Note the '+1 in front of the number. I just add the +1, like: Exten = NX/+12035551212,1,Blahblahblah ..and all is happy for me! Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] |Sent: Friday, June 03, 2005 8:16 PM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Caller ID Routing using VoicePulseConnect | |I have a question for those of you out there using VoicePulseConnect for |incoming did | |I have in my Realtime extensions Database |(the x's are replaced with my phone number) |context = voicepulse-in-01 |exten = xx/ |Priority=1 |app=NoOp |appdata = Incoming call with no callerid on xx | |However it never triggers | |I also tried using one of my other providers (voipjet for outbound) and |calling myself - i set the outbound callerid to nothing and it defaults to |202556 | |so i tried the above with the difference in |exten = xx/202556 |and this example doesn't match either! |This is the only inbound iax provider that i have. | |This is the way it's supposed to work? correct? as i have read on the wiki. | |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 ___ 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] Automatic callback feature *66
I found this before too. I am really looking for the exact opposite of this; I want the 'Calling Party' to dial *66 when they get a busy or voicemail starts. Then, some app will monitor the Called Party's channel for it to be on-hook, and then rings back the calling party to initiate the call...just like the telco can do. ...but I do appreciate the response nonetheless! Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Jay Milk |Sent: Saturday, June 04, 2005 10:28 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Automatic callback feature *66 | |Google is your friend... And so's the wiki: | |http://www.voip-info.org/wiki-Asterisk+tips+call+last+caller | | -Original Message- | From: Chris Coulthurst [mailto:[EMAIL PROTECTED] | Sent: Saturday, June 04, 2005 4:52 AM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: [Asterisk-Users] Automatic callback feature *66 | | | Does anyone have a quick-n-dirty context to implement *66 | automatic callbacks? | | I have a few people who like to have no call waiting on their | phone (can you really blame them?) It would be nice to have | something like *66, and also like 'Camp On', but instead of | waiting something like 30 seconds, monitor the channel until | it becomes available, then immediately ring back your phone | to initiate the call. | | Ok, I redundantly described what it does. Anyone have a way | to make it happen? | | Thanks much... | | Chris Coulthurst | [EMAIL PROTECTED] | | | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/aster isk-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 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] Automatic callback feature *66
I found this before too. I am really looking for the exact opposite of this; I want the 'Calling Party' to dial *66 when they get a busy or voicemail starts. Then, some app will monitor the Called Party's channel for it to be on-hook, and then rings back the calling party to initiate the call...just like the telco can do. ...but I do appreciate the response nonetheless! Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Jay Milk |Sent: Saturday, June 04, 2005 10:28 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Automatic callback feature *66 | |Google is your friend... And so's the wiki: | |http://www.voip-info.org/wiki-Asterisk+tips+call+last+caller | | -Original Message- | From: Chris Coulthurst [mailto:[EMAIL PROTECTED] | Sent: Saturday, June 04, 2005 4:52 AM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: [Asterisk-Users] Automatic callback feature *66 | | | Does anyone have a quick-n-dirty context to implement *66 automatic | callbacks? ___ 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] How to quickly replace ', ' with '|' in dialplans?
There is a perl script out there called 'sarep' which is short for search-and-replace that I use quite frequently for such things. Check it out at this URL: http://tarp.worldserve.net/software/sarep.html You can type something like: # sarep 'Sip/203' 'Zap/2r2' /etc/asterisk/extensions.conf All Sip/203s become Zap/2r2s. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Steve |Sent: Saturday, June 04, 2005 5:09 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] How to quickly replace ',' with '|' in |dialplans? | |Cool |Good to know! |I use the heck out of vi but by all means not a power user. | |Thanks for that tidbit! | |Steve | | | | | | | |On Sat, 4 Jun 2005, Jeremy McDermond wrote: | | On Jun 4, 2005, at 1:30 PM, [EMAIL PROTECTED] wrote: | | -- Messaggio Originale -- | Date: Sat, 4 Jun 2005 14:52:42 -0400 (EDT) | From: Steve [EMAIL PROTECTED] | To: Asterisk Users Mailing List - Non-Commercial Discussion | asterisk-users@lists.digium.com | Subject: Re: [Asterisk-Users] How to quickly replace ', | ' with '|' in dialplans? | Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion | asterisk-users@lists.digium.com | | | | I stink at regular expressions, but can always find what I need to get |a | | job done using google :-) | | Don't use vi (unless you figure out how to do it in vi). | I won't be much help there. | | in this case sed is your friend. | | It's a breeze to use too. | | Here's what it looks like: sed -e s/text_to_find/text_to_replace/g | inputfile outputfile | and yup you can use the same name for both files to simply update the | file. | | | You can do the same thing in vi: | | :1,$s/text_to_find/text_to_replace/g | | but sed works too. | | | Make sure you back it up first of course. | | Here's a link with some more examples using sed: | http://pegasus.rutgers.edu/~elflord/unix/sed.html | | | Hope this helps! | | Steve | | | | ___ | 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 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] Extension 'hint' info please?
I have a Polycom 500 and would love one of the line-appearance keys to show me if a certain person/people are on the phone upstairs. This 'hint' priority seems to have little-to-no documentation. So, if anyone out there has a clue about this, here are a couple of questions: Can you 'hint' Zap and IAX2 extensions? Can you concatenate extensions together? (i.e. exten = 200,hint,SIP/101SIP/202Zap/4) And the big one is. ...how does this work on a Polycom 500? Chris Coulthurst [EMAIL PROTECTED] ___ 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] 911 context, is this right?
I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten = 911,1,ChanIsAvail(Zap/1) exten = 911,2,Dial(Zap/1/911) exten = 911,3,Hangup() exten = 911,102,ChanIsAvail(Zap/4) exten = 911,103,Dial(Zap/4/911) exten = 911,104,Hangup() exten = 911,203,ChanIsAvail(Zap/5) exten = 911,204,Dial(Zap/5/911) exten = 911,205,Hangup() exten = 911,304,SoftHangup(Zap/5-1) exten = 911,305,Wait(2) exten = 911,306,Goto(204) Did I get the Priority + 101 idea right here? Chris Coulthurst [EMAIL PROTECTED] ___ 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] Teliax is DOWN
What about some sort of asterisk-level Ping app that could let one server with the app, ping the other, and check for status info, and if it doesn't like what it sees (or doesn't see anything), it would consider that channel dead? I know I'm just passing broad strokes here, but I think the idea is sound enough... Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Rich Adamson |Sent: Friday, June 03, 2005 5:50 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Teliax is DOWN | | | I wonder if the combination of qualify=yes and ChanIsAvail() | does something useful? I always meant to find out. Asterisk | does seem to monitor the outbound links and does seem to be | aware when things are down when qualify is on. | | It would be really useful, especially for those of us in remote |locations, | to have a definitive method of presenting two outgoing long | distance/international contexts and have a cmd that would choose the one |to | use based on a priority and availability. I would not want to have it | decided based only on ping time, I would want to route to my preferred | provider as long as it was reasonably good. | |The problem with attempting to do the above is 'what does down mean?' | - no registration? | - can't connect? | - connects process calls, but no ringing? | - fast busy? | |Based on past experience with multiple itsp's, it seems the majority |of the problems aren't as simple as can't connect. In the US at least, |it would seem like some form of accurate call progress detection |with timeouts would be needed in * before the above could be addressed. | | |___ |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] Call parking on Polycom 500 doesn't transfer, stays on hold
When I try and park a call by transferring to 700, it audibly says to me 701, and then instead of hanging up with me, it puts me on hold. The only way to park the call is to send it blind to 700, but then I wouldn't know which parking spot the call is in! Before I send any .conf files to the list, does anyone recognize this behavior, and have a workaround? Chris Coulthurst [EMAIL PROTECTED] ___ 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] Call parking on Polycom 500 doesn't transfer, stays on hold
Here is my features.conf [general] parkext = 700 ; What ext. to dial to park parkpos = 701-720 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 60 ; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call adsipark = yes ; if you want ADSI parking announcements pickupexten = *8; Configure the pickup extension. Default is *8 Asterisk is reporting itself as Asterisk CVS-v1-0-04/21/05-08:01:21 Software on the IP500s is: BootROM 2.6.1.0003 Sip.ld 1.5.2.0054 There is no extension listed in extensions.conf. Maybe I'm just not understanding this 'built-in' function like I should. The wiki page isn't all that detailed, so maybe its me? Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Wiley Siler |Sent: Friday, June 03, 2005 11:36 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Call parking on Polycom 500 doesn't |transfer,stays on hold | |I have IP500s and I use parking via the tranfer method without any |issue. | |Callers to my handset are sent to 701 (or other) which is played back. |The call is then dropped on my side. | |Do you have details regarding... | |What version of * you are using? |Polycom SIP and BootROM levels? | |And of course, the extension definition for 700. | |Thanks, |Wiley | | | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of Chris |Coulthurst |Sent: Friday, June 03, 2005 11:21 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] Call parking on Polycom 500 doesn't |transfer,stays on hold | |When I try and park a call by transferring to 700, it audibly says to me |701, and then instead of hanging up with me, it puts me on hold. The |only way to park the call is to send it blind to 700, but then I |wouldn't know which parking spot the call is in! | |Before I send any .conf files to the list, does anyone recognize this |behavior, and have a workaround? | |Chris Coulthurst |[EMAIL PROTECTED] | | | |___ |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 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] Call parking on Polycom 500 doesn't transfer, stays on hold
Let me just take off my stupid hat -- I figured it out. USER ERROR!! Polycom apparently has you hit Transfer, the extension to send it to (700) and send. When you hear the number of the park, you have to hit transfer again, not just hang up. Not sure why, but it works, and as long as I know how to do it, I can smile again :) Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Matthew Marlowe |Sent: Friday, June 03, 2005 1:16 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Call parking on Polycom 500 doesn't |transfer,stays on hold | |Mine works but I use Asterisk's # Transfer. Wasn't aware you can use a SIP |transfer to park a call. |- Original Message - |From: Chris Coulthurst [EMAIL PROTECTED] |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |asterisk-users@lists.digium.com |Sent: Friday, June 03, 2005 3:34 PM |Subject: RE: [Asterisk-Users] Call parking on Polycom 500 doesn't |transfer,stays on hold | | | Here is my features.conf | | [general] | parkext = 700 ; What ext. to dial to park | parkpos = 701-720 ; What extensions to park calls on | context = parkedcalls ; Which context parked calls are in | parkingtime = 60 ; Number of seconds a call can be parked | for |; (default is 45 seconds) | transferdigittimeout = 3 ; Number of seconds to wait between | digits when transfering a call | courtesytone = beep ; Sound file to play to the parked | caller |; when someone dials a parked call | adsipark = yes ; if you want ADSI parking announcements | pickupexten = *8; Configure the pickup extension. | Default is *8 | | Asterisk is reporting itself as Asterisk CVS-v1-0-04/21/05-08:01:21 | Software on the IP500s is: | BootROM 2.6.1.0003 | Sip.ld 1.5.2.0054 | | | There is no extension listed in extensions.conf. Maybe I'm just not | understanding this 'built-in' function like I should. The wiki page | isn't all that detailed, so maybe its me? | | | | Chris Coulthurst | [EMAIL PROTECTED] | | | |-Original Message- | |From: [EMAIL PROTECTED] [mailto:asterisk-users- | |[EMAIL PROTECTED] On Behalf Of Wiley Siler | |Sent: Friday, June 03, 2005 11:36 AM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: RE: [Asterisk-Users] Call parking on Polycom 500 doesn't | |transfer,stays on hold | | | |I have IP500s and I use parking via the tranfer method without any | |issue. | | | |Callers to my handset are sent to 701 (or other) which is played back. | |The call is then dropped on my side. | | | |Do you have details regarding... | | | |What version of * you are using? | |Polycom SIP and BootROM levels? | | | |And of course, the extension definition for 700. | | | |Thanks, | |Wiley | | | | | | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of Chris | |Coulthurst | |Sent: Friday, June 03, 2005 11:21 AM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: [Asterisk-Users] Call parking on Polycom 500 doesn't | |transfer,stays on hold | | | |When I try and park a call by transferring to 700, it audibly says to | me | |701, and then instead of hanging up with me, it puts me on hold. The | |only way to park the call is to send it blind to 700, but then I | |wouldn't know which parking spot the call is in! | | | |Before I send any .conf files to the list, does anyone recognize this | |behavior, and have a workaround? | | | |Chris Coulthurst | |[EMAIL PROTECTED] | | | | | | | |___ | |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 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman
[Asterisk-Users] Teliax is DOWN
Teliax is down, and can't even get to their www.teliax.com website. Anyone else having problems? Chris Coulthurst [EMAIL PROTECTED] ___ 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] Teliax is DOWN
Might be a clock off somewhere, you actually responded to this within 10 minutes of me sending to the forum... and teliax is back up again anyway..mustve been a network glitch down there in Colorado. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com |Sent: Thursday, June 02, 2005 9:24 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Teliax is DOWN | |On Thu, 2005-06-02 at 21:08 -0700, Chris Coulthurst wrote: | Teliax is down, and can't even get to their www.teliax.com website. | Anyone else having problems? | | |dont have an account with them but I am easily able to get to their |webpage. | |It is now 9:23pm PST. Took 30 minutes for your email to hit my box, so |maybe it was your isp and not teliax? | | |-- |Trixter http://www.0xdecafbad.com Bret McDanel |UK +44 870 340 4605 Germany +49 801 777 555 3402 |US +1 360 207 0479 or +1 516 687 5200 |FreeWorldDialup: 635378 ___ 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] Teliax is DOWN
When my service was down briefly, it tried for quite some time to contact the voip.teliax.com IAX server, and came back with a authentication rejected message. Now, I had this problem from here, as well as a test from another location with a completely different account with teliax, and the effect was the same. So, how to I enter in to the dialplan, to not wait for 30+ seconds to contact an unreachable server, and go on to the next-priority failover ZAP trunks? I'm thinking that something like 6 or 7 seconds is the longest I could let it wait before a dialout user thinks something's wrong and tries again (thus reentering a 7 second timeout if teliax is still down). Any suggestions? Dialplan examples? Thanks for the help.. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Chris Coulthurst |Sent: Thursday, June 02, 2005 9:29 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] Teliax is DOWN | |Might be a clock off somewhere, you actually responded to this within 10 |minutes of me sending to the forum... and teliax is back up again |anyway..mustve been a network glitch down there in Colorado. | |Chris Coulthurst |[EMAIL PROTECTED] | | ||-Original Message- ||From: [EMAIL PROTECTED] [mailto:asterisk-users- ||[EMAIL PROTECTED] On Behalf Of trixter |http://www.0xdecafbad.com ||Sent: Thursday, June 02, 2005 9:24 PM ||To: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: Re: [Asterisk-Users] Teliax is DOWN || ||On Thu, 2005-06-02 at 21:08 -0700, Chris Coulthurst wrote: || Teliax is down, and can't even get to their www.teliax.com website. || Anyone else having problems? || || ||dont have an account with them but I am easily able to get to their ||webpage. || ||It is now 9:23pm PST. Took 30 minutes for your email to hit my box, so ||maybe it was your isp and not teliax? || || ||-- ||Trixter http://www.0xdecafbad.com Bret McDanel ||UK +44 870 340 4605 Germany +49 801 777 555 3402 ||US +1 360 207 0479 or +1 516 687 5200 ||FreeWorldDialup: 635378 | |___ |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] When to use 'Answer' and when NOT to...
While everything seems to be working for the most part correctly in my mix-network of Zap and Sip phones, it occurred to me that every call, regardless of whether or not it was answered, is reporting ANSWERED in the cdr records on mysql. I was having problems with strange hang-ups the moment a call went off hook, and having Answer in the extensions.conf contexts made it all go away. Am I under-thinking the use of Answer()? Chris Coulthurst [EMAIL PROTECTED] ___ 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] ipchains for firewall, QOS howto?
I have an Asterisk PBX behind a manually-built IPCHAINS firewall machine. Can anyone tell me what I need to allow/build QOS packet rewrites through this simple NAT barrier? What do I need to pass to IPCHAINS to let QOS out to the next outside network hop? I ask this, because I have been getting intermittent jitter from my provider (TELIAX), and since it seems near-impossible to verify the source of the latency, I want to make sure I have all my Ts crossed and Is dotted before I blame something external for my issues. On the same note, what is the best way to test my connection for jitter, packet loss, etc, and still be able to determine what the potential culprit is for the problem? Thanks again, Chris Coulthurst ___ 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] Asterisk with another Asterisk
Has anyone seen a situation where, upon connecting two asterisk servers together with IAX registration, outgoing/incoming calls that route through both servers are choppy and jittery? I don't have this problem when I call out to teliax (my ITSP) directly, but if I try to make the call through the 'remote' asterisk server downtown, it gets bad. If I register my SIP phone here at home to the server downtown directly and make the call, the problem goes away again! CPU load is low, and the cable internet pipe is free and wide open with no appreciable latency. I've tried every jitterbuffer config I could think of. Any suggestions on where to find some probable causes? Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Collard Sent: Tuesday, May 31, 2005 6:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk with another Asterisk Yes, via IAX Adam Collard General Manager, ER Wireless (800) 757-5669 x4861 (810) 496-0161 Fax (517) 242-1800 Cell Nextel DC 131*256784*19 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cyril SIMON Sent: Tuesday, May 31, 2005 1:29 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with another Asterisk Hi, I'm a newbie on Asterisk and I'd like to know if it's possible to connect two or more asterisk together. In fact, I'd like install and connect some asterisk together. Thanks for advance, Cyril _ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.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 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] MWI - One mailbox, multiple extensions, lots of lights!
I am sure this has been addressed somewhere else, but I havent found it Is there a way to make multiple extensions have their MWI light flash, all for the same common voicemailbox? And to make it even trickier, what if its a mix of SIP and ZAP channels? Id like to be at my server-room desk and see MWI from my office mailbox. ___ 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] Can't transfer calls on polycom 500 after new firmware upgrade
When I try to transfer calls with a Polycom 500, (blind or not), the digits don't display properly on the screen, and the call is just put on hold. I also can't park a call for the same reasons. Is there something funny with sending digits to this polycom? I'm sure this is something silly. Sip version 1.5.2.0054 and using the old bootrom Chris Coulthurst [EMAIL PROTECTED] ___ 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
