Re: [asterisk-users] Poll: Asterisk IMAP feedback (was: Is anyonesuccessfully using IMAP storage)
We tried with MS Exchange but couldn't get it to work (MS Exchange doesn't support a master account). CP From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Thursday, October 18, 2007 11:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Poll: Asterisk IMAP feedback (was: Is anyonesuccessfully using IMAP storage) Hello, Are you using Asterisk 1.4 ? If positive, are you then successfully using IMAP storage ? Your input would be very valuable to decide if rewite of IMAP storage could be considered as bug fix (non one uses IMAP now) or as a new feature (many use IMAP storage today). So please, take a few seconds to reply as up to now (4 answers), successful IMAP user share = 0% ! Regards PS: If someone has a more effective way to gather user feedback, do not hesitate to tell. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 won't take new bootrom.ld or sip.ld...
Hi Doug, What combination of bootrom, sip version and FTP server are you using? There is a combination with vsFTPd which can cause this sort of problem. CP -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Sent: Thursday, September 27, 2007 3:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 501 won't take new bootrom.ld or sip.ld... ...even when I do the factory reset (4-6-8-* then 456). I tried using FTP and TFTP, but even though the phone uploads the log, I get these errors: 0927211350|app1 |3|00|Time has been set from 0.us.pool.ntp.org(69.60.124.59). 0927211350|cfg |4|00|Could not get all 512 bytes of the header. 0927211351|cfg |4|00|Could not get all 512 bytes of the header. 0927211422|app1 |4|00|Loaded application sip.ld successfully, errors 0x20. 0927211422|app1 |6|00|Uploading boot log, time is THU SEP 27 21:14:22 0927211422|2007 Has anyone seen this before? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP 4000 Soundstation SIP Conference PhoneQuestion
Hi Matt, We have one and it works very well - usual Polycom quality, as others have attested. The only thing we have noticed is a reluctance to download its config files via FTP when using a VLAN tag. CP Matt wrote: Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CID on Polycom Phones
Hi David, Disable URL dialing (url-dialing in the feature/ section of sip.cfg. CP Klaverstyn, David C wrote: Hi All, I have a site using Polycom 501 phones and for some reason the caller ID of the phone number is coming up as sip:number@ip of server Does anyone know why? It seems to be a Polycom thing as a Linksys phone displays the CID number as just the number. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unsubscribe
And yet, it's shorter than your HTML/image-ridden sig. :-) CP Wiley Siler wrote: Disclaimer at the bottom still looks ridiculous even in Spanish… LOL *Wiley E. Siler **Director of Information Technology* 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] www.education2020.com http://www.education2020.com/ cid:image003.jpg@01C77AC4.A558AFE0 Helping students on a mission. Graduation and beyond. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of [EMAIL PROTECTED] *Sent:* Friday, May 18, 2007 4:43 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] unsubscribe Cristian López F. Integración y Tecnología - Terra Chile Phone: (56 2) 330 6966 movil: 56-92401759 E-mail: [EMAIL PROTECTED] Este correo y su contenido solamente interesan a las personas autorizadas de TERRA NETWORKS CHILE. Si usted fue receptor de este correo por error, por favor no lo tome en cuenta y avise al remitente. This message is solely of the interest of TERRA NETWORKS CHILE or its businesses. If you have received this e-mail by error, please ignore it and notify the sender. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?
I use Bluefish, and have developed a syntax-highlighting template for Asterisk conf files, if you're interested. CP Steve Finkelstein wrote: This might be of some assistance: http://www.voip-info.org/wiki/view/vim+syntax+highlighting - sf Olivier wrote: Hi, New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor with which I could easily edit Asterisk config files. It seems Kate provide this type of service but I couldn't find anything specific to Asterisk (unlike vim) What's your advice ? Best regards !DSPAM:1020,464b158e638175802679531! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:1020,464b158e638175802679531! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail volume
Try the 'g' option to VoiceMail(). CP Stephen Bosch wrote: Hi: I have a user saying that the volume of voice mails is too low. Is there a way to tweak the recording level for voice mail? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail on Different Server
That's the way we want to go, but have been unable to divine the correct settings for getting it working with MS Exchange. CP Tim Panton wrote: If I were starting a project now, I'd take a look at the (newish) support for IMAP storage for voicemail. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail on Different Server
mount -o intr,nolock ought to do the trick. we're using those options now, but thankfully haven't had reason to find out if they work or not yet. CP Doug Garstang wrote: No, you can get Asterisk and NFS to work fine together. It was in my past job, so I can't remember the exact settings, but there was some magic combination of NFS client mount settings that would cause Asterisk to return immediately, rather than hang, if there was an NFS communications problem. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail on Different Server
It will stall asterisk - ask me how I know.. :-) CP Gordon Henderson wrote: On Tue, 24 Apr 2007, Forrest Beck wrote: I've heard there are problems using NFS as a storage device.??? I've used NFS for many many years on 100s, maybe 1000s of servers in this time. It's great. Just works and does exactly what it says on the tin. I use it at home, for my clients, on my hosted servers, everywhere. (well, almost!) BUT... If the NFS server should go offline for whatever reason then the client systems that are reading/writing the data will stall, and depending on how you've got them setup they will stall hard and not continue until the server returns. I haven't tried it with asterisk yet, so I do not know what will happen to the voicemail system should the NFS server go offline for whatever reason. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Voicemail on Different Server
Why not export an NFS mount from one server to the other? That's what we do. CP -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck Sent: Tuesday, April 24, 2007 5:28 PM To: Asterisk Users List Subject: [asterisk-users] Voicemail on Different Server I have two seperate systems at two different locations. Each hosts there own voicemail for their phones. I have thought about just having all voicemail on one server. Is the best way to do this just through a dial app? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk (1.4) and hints/presence/BLF
Hi John, Try 1.4.2 - there was a bug in earlier versions that produced the symptoms you describe (http://bugs.digium.com/view.php?id=8848, and various related ones). A. John Hughes wrote: Playing with hints/presence/BLF on asterisk I've made the following discoveries. 1. The wiki at http://www.voip-info.org/wiki/view/Asterisk+presence says: If you add incominglimit=1 to your peer in sip.conf, the SIP channel will notify you when that extension is busy. As incominglimit is obsolete you can use call-limit. Also you don't need to limit it to one, just having a call-limit at all works. (Tried with call-limit 20). What is the logic behind the linking of presence to call-limit? 2. A phone is only seen as busy if it's received an incoming call. Outgoing calls don't change the state. Why? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is your Backup Strategy?
Hi Forrest, This is very close to the option we use - we have 2 identical servers, and keep the configs the same using Visual Sourcesafe to shadow all changes on server1 to server2. Voicemail and other lib files are stored on the spare server and exported to the primary server using NFS. If the primary server fails, DNS SRV records have the Polycom phones failover to the spare server. We physically move the PRIs from one server to the other. Forrest Beck wrote: 2) Have two servers with the same dialplan. One in each location. Each server has it's own TDM cards installed. Phones on Site A will register with the server on Site A, and phones on Site B will register with the server on Site B. Then using Polycom phones, they will failover to using the server not on their site, if their primary isn't available. I have setup scripts to copy the dialplan from one server to the other then reload asterisk nightly. The biggest Con to this is I have to be sure my dialplans don't get different. The user's voicemail wouldn't be available until their primary server is back up, but that's OK. -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IMAP Voicemail with MS Exchange
Hi there, We're trying to get IMAP voicemail storage working on an MS Exchange server - I would be grateful if anyone who has successfully done this could post the magic soup here, as extensive Google searching has yielded nothing other than tantalizing references to it being done without any specifics. -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemails with occasional speeded up portions
Greetings, Very occasionally, we have a complaint from a user that a portion of a voicemail message is very speeded up - like when you press the fast- forward button on an old-fashioned tape dictaphone. This affects both the server-stored and emailed copies of the message. I have a sample if anyone is interested. Has anyone else experienced this? CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inserting a pause with Sipura in between
We have it working fine on an SPA-3000. CP On Feb 5, 2007, at 10:42 PM, Joseph wrote: I've a problem with inserting a pause and dialing additional numbers when going through Sipura-3000 exten = _12,1,Dial(SIP/[EMAIL PROTECTED],30,D(ww18)) D() doesn't work as it sends the DTMF tones right after FXS connects to FXO; though, I want insert a pause and send additional numbers after connection goes through FXO. Is it possible? -- #Joseph ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Polycom buddy status
Hi there, We traced this issue to transfers (see http://bugs.digium.com/ view.php?id=8848). On Monday, I will be attaching the various debugs to the case as requested. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 26-Jan-07, at 5:16 PM, James Fromm wrote: Olle E Johansson wrote: 26 jan 2007 kl. 16.31 skrev James Fromm: Olle E Johansson wrote: 24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling: James Fromm wrote: The behavior we see is that the SIP interface in the queue will sometimes not release from the in-use state. Connecting to the interface from another SIP device and immediately hanging up will clear the state. The phones in question are configured with one line that will except only one call. The device itself does not think it is in-use because it will accept another call. Something in the SIP channel driver is not clearing the state when a call is completed. There is definitely no correlation between this and Asterisk restarting. In fact, if a device is 'stuck' on in-use, restarting Asterisk will clear the state. I've been working on this for a week now. It only started for us because I just implemented the call-limit option in the sip.conf in Asterisk for the devices. See my posts with subject 'Queue and Interface time out'. I believe there is/was a bug relating to call-limit. Buddy Watch doesn't work if you use call-limit and if a call from a queue is transfered, the call-limit is not released until the original call is terminated. I do not know if these issues have been fixed or not. Again, a relation to call transfer. I think the bug is that we don't handle call-limits properly during a call transfer. That needs to be verified and fixed. There may be, but transfers are not the cause of the issue I describe. SIP interfaces that are members of a Queue, will erratically not be released from 'in-use' when a call is completed. I have tested with both caller terminated and agent terminated calls and both will cause this behavior. It happens on approximately 20% of all calls the queue members receive. Dialing the SIP device with another device will immediately free the status. I wonder if this only happens on calls sent to the SIP device by the Queue application. I will test that today. If you are using chan_agent as a proxy channel, check if that changes things. We don't have agents defined so I don't think chan_agent applies. The Queue's members are assigned through the management port from an application running on the the agent's PC. I think the Queue application loses sync to the SIP channel driver's information containing the state of the SIP interfaces. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Bootcamp in Pacific Northwest (Vancouver, BC)
Greetings, The District of North Vancouver, a municipal government in BC, Canada, is hosting a Digium instructed Asterisk Bootcamp at our training center from February 5th-9th, 2007. Primarily arranged to provide training to some of our staff, there is space available for others to avail of this opportunity to obtain Asterisk bootcamp training in the Pacific Northwest. Space on the course can be booked via the Digium web site at http://www.digium.com/en/training/locator/enroll/46. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk-users Digest, Vol 29, Issue 71
Here's how to unsubscribe: First, ask your Internet Provider to mail you an Unsubscribing Kit. Then follow these directions. The kit will most likely be the standard no-fault type. Depending on requirements, System A and/or System B can be used. When operating System A, depress lever and a plastic dalkron unsubscriber will be dispensed through the slot immediately underneath. When you have fastened the adhesive lip, attach connection marked by the large X outlet hose. Twist the silver-coloured ring one inch below the connection point until you feel it lock. The kit is now ready for use. The Cin-Eliminator is activated by the small switch on the lip. When securing, twist the ring back to its initial condition, so that the two orange lines meet. Disconnect. Place the dalkron unsubscriber in the vacuum receptacle to the rear. Activate by pressing the blue button. The controls for System B are located on the opposite side. The red release switch places the Cin-Eliminator into position; it can be adjusted manually up or down by pressing the blue manual release button. The opening is self-adjusting. To secure after use, press the green button, which simultaneously activates the evaporator and returns the Cin-Eliminator to its storage position. You may log off if the green exit light is on over the evaporator. If the red light is illuminated, one of the Cin-Eliminator requirements has not been properly implemented. Press the List Guy call button on the right of the evaporator. He will secure all facilities from his control panel. To use the Auto-Unsub, first undress and place all your clothes in the clothes rack. Put on the velcro slippers located in the cabinet immediately below. Enter the shower, taking the entire kit with you. On the control panel to your upper right upon entering you will see a Shower seal button. Press to activate. A green light will then be illuminated immediately below. On the intensity knob, select the desired setting. Now depress the Auto-Unsub activation lever. Bathe normally. The Auto-Unsub will automatically go off after three minutes unless you activate the Manual off override switch by flipping it up. When you are ready to leave, press the blue Shower seal release button. The door will open and you may leave. Please remove the velcro slippers and place them in their container. If you prefer the ultrasonic log-off mode, press the indicated blue button. When the twin panels open, pull forward by rings A B. The knob to the left, just below the blue light, has three settings, low, medium or high. For normal use, the medium setting is suggested. After these settings have been made, you can activate the device by switching to the ON position the clearly marked red switch. If during the unsubscribing operation you wish to change the settings, place the manual off override switch in the OFF position. You may now make the change and repeat the cycle. When the green exit light goes on, you may log off and have lunch. Please close the door behind you. On Dec 19, 2006, at 2:22 AM, [EMAIL PROTECTED] wrote: Hi, I want to unsubscribe from asterisk-users-request-lists, and donot want to recieve mail any more. Kindly unsubscribe me... sanchal singh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom IP4000 and vsftpd 2.0.1
Is anyone else having trouble getting a Polycom IP4000 (running SIP 1.6.7 and BootROM 3.1.3) to download its configuration files from a vsftpd 2.0.1 server? We have 100+ IP501s that manage this without problems, but the IP4000 keeps timing out. We have opened a case with Polycom, but they are insisting that it is our configuration files that are at fault, even though the phone times out on bootrom.ld, long before it attempts to load the configuration files. I did turn up some postings about IP501s, BootROM 3.1.3 and vsftpd 2.0.3, and wonder if this might be a similar issue. CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Low beep on voicemail
Just 'sox -v 1.5 beep.gsm loudbeep.gsm' ? CP On 2-Dec-06, at 11:29 AM, Peder @ NetworkOblivion wrote: We've had a few people complain that the beep before leaving a voicemail is not loud enough and too short. Does anybody have a recorded beep that they can share, that is a little louder and a little longer? We've had this box in production for 2+ years, so I hate to mess with the gain on the PRI or anything like that because everything else works fine. I know nothing about recording sounds, and I am sure I could spend a few hours and come up with a suitable version, but I thought I'd ask around before I waste my time trying to figure it out. Thanks in advance. Peder ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4beta3 help
IIRC, menuselect requires ncurses-devel (or your distro's equivalent). CP On Dec 1, 2006, at 7:05 AM, Doug Crompton wrote: No, no menuslect on system beside * I unzipped it, ran configure, then make (or make menuselect) they both give the same immediate error 3. From what I see with 1.4.x it might be good to have a completely seperste list. I suspect there will be tons of email volume once it's use or attempt of use ramps up! Doug On Fri, 1 Dec 2006, Tim Panton wrote: On 1 Dec 2006, at 03:49, Doug Crompton wrote: no - make menuselect - does the same thing. Have you got a (non asterisk) binary or shell script called menuselect in your path? try which menuselect Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * [EMAIL PROTECTED] * * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FS: Sangoma 10 port FXO card
Please don't cross post FS items to *-users - that's what *-biz is for. CP On Nov 24, 2006, at 10:45 AM, Mark Phillips wrote: Hi all, I have a surplus Sangoma 10 port FXO card for sale. This model could be upgraded to 12 ports or even changed to FXS or a combo of FXO/FXS by changing the grand-daughter cards (each card supports 2 lines). You could also downgrade the card by removing any or all of the daughter cards. I'm asking US$450 plus shipping to the lower 48. Paypal or Master/Visa only. Thanks Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel error
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 CP On Nov 22, 2006, at 8:40 PM, ram wrote: Hi where can i buy that Book Ram On 11/22/06, Patrick [EMAIL PROTECTED] wrote: On Wed, 2006-11-22 at 15:45 +0530, ram wrote: [snip] Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring switchtype Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring signalling Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring rxwink [snip] is this error cause any problem or just ignore this ^ Error? Where does it say error? Read the messages carefully and you will see that it says.. WARNING. If it was an error it would have said ERROR. But it didn't. Phew. Just a harmless warning. And to figure out what the warnings mean, I suggest you buy/get the Asterisk book. It's very helpful to learn about these basic things. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls from asterisk
Just use Set(CALLERID(name)) in your dialplan - that's what we do. CP On Nov 23, 2006, at 12:00 AM, Eric Bishop wrote: When we have calls that originate click-to-daial apps that use the manager interface they always originate from asterisk is there any way to change the from name? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms
Thanks, John - this confirms what we are seeing. 'show hints' output isn't changing, so it looks like a bug. I'll open one and see what happens. A. On Nov 21, 2006, at 5:44 PM, John Lange wrote: Hints are not working in 1.4b3 period. Snom 360s do not show any status updates. However, before you file a bug report you might want to check to see if there are changes to the way hints are implemented in 1.4. It might be a configuration problem rather than a bug but I have not had time to look into it. John On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote: Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip debug' shows absolutely no NOTIFY or XML presence messages as calls progress.. Next stop Mantis :-) CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms
http://bugs.digium.com/view.php?id=8405 On Nov 22, 2006, at 9:11 AM, Anthony Rodgers wrote: Thanks, John - this confirms what we are seeing. 'show hints' output isn't changing, so it looks like a bug. I'll open one and see what happens. A. On Nov 21, 2006, at 5:44 PM, John Lange wrote: Hints are not working in 1.4b3 period. Snom 360s do not show any status updates. However, before you file a bug report you might want to check to see if there are changes to the way hints are implemented in 1.4. It might be a configuration problem rather than a bug but I have not had time to look into it. John On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote: Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip debug' shows absolutely no NOTIFY or XML presence messages as calls progress.. Next stop Mantis :-) CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing from Placed Calls on Polycom IP501 doesn't always work
We narrowed this down to when the 'New Call' softkey was used to initiate the call. When this key was used, the corresponding 'Placed Calls' entry wouldn't work. Any other method of placing the call does work. An upgrade to 1.6.7 fixes the issue. CP On Nov 16, 2006, at 4:34 AM, John Marvin wrote: Noah Miller wrote: I never ran 1.6.6 for any length of time. 1.6.7 and 2.0.1 don't seem to suffer this issue. 2.0.1 has some buddy watch problems, so you may not want to use it, but 1.6.7 should be OK. I've been running 1.6.6 for quite a while, and I have been quite annoyed by this bug. However, the release notes for 1.6.7 did not mention fixing this problem, so I did not have any motivation for upgrading. But, since you said that you did not see the problem on 1.6.7 I decided to upgrade and see if the problem was fixed. It appears to have fixed it, although I can't be sure yet, because sometimes a call placed from the placed calls list did work on 1.6.6, so I don't have enough of a sample size yet to be sure the bug is gone. I sure hope it is. Thanks for the info. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing from Placed Calls on PolycomIP501doesn't always work
Thanks, Noah - we'll try 1.6.7 and see if the problem goes away. CP On 15-Nov-06, at 11:55 AM, Noah Miller wrote: Has anyone noticed that attempting to place a call from the Placed Calls list on a Polycom IP501 by pressing the 'Dial' softkey sometimes simply returns the phone to the idle screen? Yes, I've seen it. We're running 1.6.6, what firmware version do you have? We're running SIP 1.6.6.0036 on the 3.1.3.0131 BootROM. Did you come up with any reason/fix for this? I never ran 1.6.6 for any length of time. 1.6.7 and 2.0.1 don't seem to suffer this issue. 2.0.1 has some buddy watch problems, so you may not want to use it, but 1.6.7 should be OK. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialing from Placed Calls on Polycom IP501doesn't always work
Hi James, We're running SIP 1.6.6.0036 on the 3.1.3.0131 BootROM. Did you come up with any reason/fix for this? CP On Nov 13, 2006, at 11:00 PM, James Andrewartha wrote: Anthony Rodgers wrote: Greetings, Has anyone noticed that attempting to place a call from the Placed Calls list on a Polycom IP501 by pressing the 'Dial' softkey sometimes simply returns the phone to the idle screen? It is not related to the number being dialed, as we have observed two entries for the same number, one of which worked and the other didn't. We've experimented with calls that weren't answered at all, calls that were terminated by the caller and calls terminated by the recipient with no discernible pattern. Yes, I've seen it. We're running 1.6.6, what firmware version do you have? -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom - how to 'buddy watch' trunks?
Have you tried setting up a hint for a ZAP channel? exten = foo,hint,ZAP/bar Then make a directory entry for foo in your Polycom directory for foo - just as you would if the hint was for a SIP channel. CP On Nov 14, 2006, at 4:26 AM, Robert Jenkins wrote: Hi, I've recently got some Polycom 501 601 phones. I have buddy watch working showing the status of users listed in the directory. I would like to also have the status of the trunks (ZAP via TDM2400E SIP) on the IP601 Sidecar display, but I cannot so far find any info on this? Thanks, Robert. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialing from Placed Calls on Polycom IP501 doesn't always work
Greetings, Has anyone noticed that attempting to place a call from the Placed Calls list on a Polycom IP501 by pressing the 'Dial' softkey sometimes simply returns the phone to the idle screen? It is not related to the number being dialed, as we have observed two entries for the same number, one of which worked and the other didn't. We've experimented with calls that weren't answered at all, calls that were terminated by the caller and calls terminated by the recipient with no discernible pattern. Regards, CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unsubscribe
Here's how to unsubscribe: First, ask your Internet Provider to mail you an Unsubscribing Kit. Then follow these directions. The kit will most likely be the standard no-fault type. Depending on requirements, System A and/or System B can be used. When operating System A, depress lever and a plastic dalkron unsubscriber will be dispensed through the slot immediately underneath. When you have fastened the adhesive lip, attach connection marked by the large X outlet hose. Twist the silver-coloured ring one inch below the connection point until you feel it lock. The kit is now ready for use. The Cin-Eliminator is activated by the small switch on the lip. When securing, twist the ring back to its initial condition, so that the two orange lines meet. Disconnect. Place the dalkron unsubscriber in the vacuum receptacle to the rear. Activate by pressing the blue button. The controls for System B are located on the opposite side. The red release switch places the Cin-Eliminator into position; it can be adjusted manually up or down by pressing the blue manual release button. The opening is self-adjusting. To secure after use, press the green button, which simultaneously activates the evaporator and returns the Cin-Eliminator to its storage position. You may log off if the green exit light is on over the evaporator. If the red light is illuminated, one of the Cin-Eliminator requirements has not been properly implemented. Press the List Guy call button on the right of the evaporator. He will secure all facilities from his control panel. To use the Auto-Unsub, first undress and place all your clothes in the clothes rack. Put on the velcro slippers located in the cabinet immediately below. Enter the shower, taking the entire kit with you. On the control panel to your upper right upon entering you will see a Shower seal button. Press to activate. A green light will then be illuminated immediately below. On the intensity knob, select the desired setting. Now depress the Auto-Unsub activation lever. Bathe normally. The Auto-Unsub will automatically go off after three minutes unless you activate the Manual off override switch by flipping it up. When you are ready to leave, press the blue Shower seal release button. The door will open and you may leave. Please remove the velcro slippers and place them in their container. If you prefer the ultrasonic log-off mode, press the indicated blue button. When the twin panels open, pull forward by rings A B. The knob to the left, just below the blue light, has three settings, low, medium or high. For normal use, the medium setting is suggested. After these settings have been made, you can activate the device by switching to the ON position the clearly marked red switch. If during the unsubscribing operation you wish to change the settings, place the manual off override switch in the OFF position. You may now make the change and repeat the cycle. When the green exit light goes on, you may log off and have lunch. Please close the door behind you. CP On Nov 9, 2006, at 8:01 AM, Adam Mattina wrote: Adam Mattina Networking Systems Support Layer 8 Group, Inc. 585.442. [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nortel Option 11C and SIP gateway integration
Hi Heison, We got our 11C working using a direct PRI connection to a Digium card in our Asterisk server. We ended up having to use two PRIs to get CallerID working properly: calls to the Nortel from Asterisk are on a 5ESS call-by-call trunk, which effectively has the Nortel treat the Asterisk server like a CO; calls to Asterisk from the Nortel are on an NI2 tie-trunk to allow the Nortel to send CallerID to the Asterisk server. Hope this helps - I have the Nortel config we used in a PDF if you need it. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 3-Nov-06, at 6:21 AM, Heison Chak wrote: Hi, We have a Nortel Option 11C (with Succession 3.0), with 3 PRI cards connected to: 1. PSTN 2. ITG network to our other 2 offices on a 4-digit dialplan 3. SIP media gateway (for Asterisk) We normally dial access code 9 for outside PSTN calls, and when the SIP media gateway was introduced, a new access code 8 was created. Inbound calls from Nortel (originating from the PSTN, from any office handset) are being delivered to the PRI trunk on the SIP media gatway then onwards to Asterisk. However, any outgoing calls made from Asterisk, into Nortel via SIP gateway is being rejected. To narrow down the possibility, we have tried 2 different SIP gateways - AudioCodes Mediant 1000 and Cisco AS5300, and they both exhibit the same behavior (incoming works fine, ALL outgoing calls are being rejected). Attached is the capture of the console message on the Nortel side while an outbound call was made. Calls from x1567 (Cisco 7960 registered to Asterisk) to x1500 (digital extension on Option 11C) is being reject with CAUSE #21. The capture also shows a successful inbound call while 4169771414 (digital handset on Opt 11C) called x1695 (Meetme on Asterisk) via the same PRI card (Ch. 4 23) was completed with release cause #16. We suspect there is some authorization code or ACL that needs to be put in place, so that calls made to the Opt 11C can be routed. We have tired talking to 3 local Nortel vendors, AudioCodes and none has been able to help us rectify this issue. We are looking for someone who can help us identify what the problem is so that we can get this working. Thanks -Heison -- Heison ChakEmail: [EMAIL PROTECTED] 14 Bartlett Rd.Phone: +1 905 887 4694 x1508 Markham, ON L6C 2Y6Toll: +1 888 887 4694 x1508 Canada Cell: +1 416 417 8893 Fax: +1 905 887 4694 UK:+44 0207 099 5883 HK:+852 3596 4261 soma call fail signature.asc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX to a Nortel MICS PBX
Can you be more specific? What sort of linkages are available between the two offices? CP On 22-Oct-06, at 10:38 PM, dthurn wrote: What's the best way to connect an Asterisk PBX to a Nortel MICS PBX. I have two offices that I want to link together. TTFN ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vancouver Asterisk User Group
Greetings, This is my annual post-Astricon attempt to start an Asterisk User Group in the Vancouver, BC, area. If you are interested, please reply off-list. Regards, -- Anthony Rodgers (CunningPike) Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Enterprise Asterisk User Group
Greetings, This is my annual post-Astricon attempt to get an Enterprise Asterisk User Group off the ground. We are a municipal government using Asterisk to replace a legacy PBX. I'd be interested in starting a group of similar enterprise users (say, 100 seats or more) other than resellers, carriers and call-centers who are using Asterisk to support their non-telecom-related business - I don't envisage any geographical limitation to the group (there seem to be few enough of us as it is!). If you are interested, please let me know off-list. Regards, -- Anthony Rodgers (CunningPike) Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need help using tftp for polycom 501
IMHO, FTP really is the way to go - you get the ability to have the phones detect config file changes and automatically reboot, and you get the ability to upload logs, custom configs and directories from the phones. We use vsftpd, with the default user and password for the phone. CP On 25-Oct-06, at 7:29 AM, Doug Lytle wrote: Marlin Unruh wrote: Glad to say I got it working. Sad to say I had to go to Windows to accomplish it. I used tftpd32 and it worked perfect. I would like to use tftp under Linux. May I will try again later. Why not use just standard FTP? I use ProFTP and setup a Polycom user. Works great. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SP4000 ftp problem
We had/have this problem, too - we eventually got it working (just by constantly rebooting it), but it seems that something's not working properly somewhere.. Can you look in your phone's boot log and see if you are getting any errors? We were seeing errors relating to the phone not being able to read sip.ld properly. CP On 23-Oct-06, at 5:51 PM, Edwin Lam wrote: i recently bought an SP4000 conference phone but having problem provisioning it using ftp, every time it just hangs at Updating initial configuration... screen. when i switch it to tftp, it'll work fine. i though it was bootrom/firmware issue so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the setting works fine w/ SP501 with bootrom 3.1.2/sip 1.6.3 -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Becoming a User on IRC
Hi Eddie, Connect to irc.freenode.net, and then type this: /msg nickserv register password nickserv will tell you that your nick is now registered. Then type this: /j #asterisk Say hi to CunningPike when you get there. CP On 24-Oct-06, at 1:12 PM, Eddie Johnson Jr wrote: Hello Dovid, My firsts time doing this what is MOTD? I also tried what you suggested /msg #asterisk username register and it did not work. I must not be doing something correct because I had a couple of other people try and not successful. Any suggetions? Ed From: [EMAIL PROTECTED] [mailto:asterisk- [EMAIL PROTECTED] On Behalf Of Dovid B Sent: Tuesday, October 24, 2006 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Becoming a User on IRC You cant PM anyone if you arent registerd. When you message nickserv copy exaclty how it is written in the MOTD (except the password part). - Original Message - From: Eddie Johnson Jr To: asterisk-users@lists.digium.com Sent: Tuesday, October 24, 2006 2:13 PM Subject: [asterisk-users] Becoming a User on IRC Hello, I followed the directions for setting up a user on Asterisk IRC. I type the following: /msg #asterisk username register password /msg #asterisk set alternative username And I get /msg Nick Serv help register. I messaged the moderator a couple of times to no avail. What am I do wrong? Thanks, Ed ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASterisk Start problem
Did you compile and install these in the correct order: zaptel libpri asterisk CP On 23-Oct-06, at 5:47 AM, ram wrote: Hi all I have installed 1.2.12.1 in FC5 with libpri.1.2.4 when i start iam getting the following error and it quits == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 __load_resource: libpri.so.1.0: cannot open shared object file: No such file or directory Oct 23 16:16:07 WARNING[11084]: loader.c:554 load_modules: Loading module chan_zap.so failed! [EMAIL PROTECTED] agc]# Ouch ... error while writing audio data: : Broken pipe what is the problem, any suggestions ? Ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FOP run control for CentOS/RHEL
Like the one that comes with it?
[EMAIL PROTECTED] ~]$ sudo more /etc/init.d/op_panel
#!/bin/bash
#
# chkconfig: 2345 99 15
# description: Flash Operator Panel
# processname: op_server.pl
# source function library
. /etc/rc.d/init.d/functions
DAEMON=/usr/local/op_panel/op_server.pl
OPTIONS=-d
RETVAL=0
case $1 in
start)
echo -n Starting Flash Operator Panel:
daemon $DAEMON $OPTIONS
RETVAL=$?
echo
[ $RETVAL -eq 0 ] touch /var/lock/subsys/op_server.pl
;;
stop)
echo -n Shutting dows Flash Operator Panel:
killproc op_server.pl
RETVAL=$?
echo
[ $RETVAL -eq 0 ] rm -f /var/lock/subsys/op_server.pl
;;
restart)
$0 stop
$0 start
RETVAL=$?
;;
reload)
echo -n Reloading Flash Operator Panel configuration:
killproc op_server.pl -HUP
RETVAL=$?
echo
;;
status)
status op_server.pl
RETVAL=$?
;;
*)
echo Usage: op_panel {start|stop|status|restart|reload}
exit 1
esac
exit $RETVAL
CP
On 16-Oct-06, at 1:08 AM, Eric Bishop wrote:
Anyone have a sane rc script for FOP on CentOS/RHEL systems?
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Re: [asterisk-users] How big is *your* dialplan??
Local government office with approximately 100 sets (going to 600): 593 extensions (1241 priorities) in 88 contexts CP On 10-Oct-06, at 1:16 PM, Steve Murphy wrote: Hello! In my relentless quest for knowledge, I pose this question: who's got the biggest dialplans, and how big are these monsters? What's the biggest dialplan in use right now? If you feel you are a competitor, let me know how many contexts/extensions/priorities you are dealing with. Maybe the context with the most extensions, the extension with the most priorities would be interesting... For example: Digium's dialplan is roughly 50 contexts, 304 total extensions, 870 total priorities. My home system has 100 contexts, 400 total extensions, 935 total priorities. My biggest extension has 129 priorities... no inflation or useless cruft there, either... mostly. These would seem small compared to some dialplans out there, I'll bet. murf -- Steve Murphy Software Developer Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware?
Hi Eric, Here's all we had to do: 1. Make sure that the 'Presence' feature is enabled in your phones: feature feature.1.name=presence feature.1.enabled=1.. in sip.cfg (or maybe ipmid.cfg, depending on the age of your SIP application) 2. Create a hint priority in extensions.conf for the extension whose status you want to monitor: exten = 2348,hint,SIP/2348-OfficeSIP/2348-Kitchen Remember that a) the hint priority does not seem to support wildcard matching b) the SIP/2348 should actually represent the SIP channel(s) that you want to monitor (this may not be the same as the extension) and c) channels can be combined using '' just like a Dial() command to monitor the status of more than one channel. 3. Set up a directory entry on the phone that you want to use to monitor the extension above. Set the 'Watched buddy' property of the directory entry to 'Yes', and the Location property to the extension that you used above (2348 in this case). That's it - let me know if you have trouble getting it work. CP On 2-Oct-06, at 11:34 PM, Eric Bishop wrote: Does anyone have an end-to-end summary of how they have successfully set up the buddy feature including all the relevant Asterisk and Polycom config snippets. All I have been able to do so far is scrounge up bits and peices from the list and Wiki - nothing that covers the entire process... I think a lot of people would benefit from that (myself included)... On 10/3/06, Paul Dugas [EMAIL PROTECTED] wrote: Install went fine. No troubles other than this and it'd be minor if one of the reasons for the update wasn't to expand the number of buddies allowed on the IP601+sidecards we're adding for the attendant. Ugh... Anyway, directory entries haven't changed: ?xml version=1.0 standalone=yes?^M !-- $Revision: 1.2 $ $Date: 2004/12/21 18:28:05 $ --directory item_list item lnDoe/ln fnJane/fn ct1001/ct sd1/sd bw1/bw /item /item_list /directory The config entries you referred to are set in my global sip.cfg and apply to all of the units. Looks right to me. Did some sniffing and Asterisk is sending a NOTIFY like so: ... ?xml version=1.0 encoding=ISO-8859-1? presence xmlns=urn:ietf:params:xml:ns:pidf xmlns:pp=urn:ietf:params:xml:ns:pidf:person xmlns:es=urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status xmlns:ep=urn:ietf:params:xml:ns:pidf:rpid:rpid-person entity=sip:[EMAIL PROTECTED] pp:personstatus /status/pp:person noteReady/note tuple id=1001 contact priority=1 sip:[EMAIL PROTECTED]/contact statusbasicopen/basic/status /tuple /presence --- Extension Changed 1001 new state Idle for Notify User x1002 pbx*CLI Hmmm On Mon, 2006-10-02 at 22:14 -0400, Scott Higginbotham wrote: I did the same thing with the Polycom's - upgraded all mine from 1.6.x to 2.0.1 but I had great success and no problem with the buddy watch / presence feature --- if anything, it works a little better. Whats your mac-address-directory.xml configuration file look like? Did you make any changes to the mac-address-phone.cfg file? do you have the line of: up.useDirectoryNames=1 feature.1.name=presence feature. 1.enabled=1 In the config? Scott Higginbotham Systems / Network Operations Manager 215.259.2185 or 1.800.835.5710 ext 2185 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Dugas Sent: Monday, October 02, 2006 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom Buddy Watch Broken with 2.0.1 Firmware? I updated a batch of Polycom IP501 phones and an IP601 to the 2.0.1 firmware to get the new NAT keep-alive feature and the ability to watch more than a handful of buddy contacts but it appears to have broken the buddy-watch feature. Is anyone seeing this? Anybody know if it's a Polycom problem or something on the Asterisk end? I'm running a recent (2 days ago) copy of the 1.2 trunk. In a rather bone-headed move, I updated the firmware and Asterisk at the same time so I'm unable to tell which is the culprit. Curious, Paul -- Paul Dugas, Computer EngineerDugas Enterprises, LLC [EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA -- This e-mail and any attachments are confidential. If you receive this message in error or are not the intended recipient, you should not retain, distribute, disclose or use any of this information and you should destroy the e-mail and any attachments or copies. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P
Likewise, Ronnie, we have 2 PRIs going to an 11C - let me know if I can help. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Sep 27, 2006, at 2:42 PM, Savoy, Kevin - Williston, ND wrote: Ronnie I have 4 non-PRI’s connected to a Nortel 11C and I had played with PRI connections before and got them working. If you want to call me we can go over your set up and compare with mine. Kevin Savoy 701-774-4023 Novo1 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronnie Jones Sent: Wednesday, September 27, 2006 2:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] RE:T1 timing errors Nortel 61C with TE110P I have no experience on the Nortel side, but will comment on the timing thingie. The asterisk T1 card (port going to the Nortel) will always generate T1 timing on the transmit side of the T1. There is no way to turn it off (by T1 Spec's). So, letting the Nortel use CLOK = EXT is perfect. The sync parameter in /etc/zaptel.conf for that same T1 port should probably be set to zero, but that statement is somewhat dependent on what the other ports on the Asterisk T1 card are used for. If there are no other Asterisk T1 card ports in use, then I'd suggest setting the sync parameter to 1. If at least one other Asterisk T1 port is in use and goes to a central office, then turn that port's sync to 1 and the Nortel port sync to 0. (Keep in mind the digium T1 cards only have one clock on board, and syncing that clock to a T1 coming from a central office is the right thing to do. Once that clock is in sync, then the Nortel will sync to asterisk.) I'm a little confused with your last paragraph when you say the circuit does establish and pass calls but resets frequently due to slips. Are those calls to/from asterisk talking to the Nortel? Yes that is correct. The Nortel switch connects to the PSTN but not the Asterisk. It connects to the Nortel. While the circuit is up I can call extensions on the Nortel from the Asterisk and visa versa. Or, are you routing incoming pstn calls from the central office through asterisk to the Nortel? No Also, have you tried any of the pri show ... commands in asterisk, or any of the pri debug items? Yes. When the circuit is up I can pri show span 1 and it show partitioned up and active. Ronnie Jones Engineer - ICT Clay Electric Cooperative, Inc 352-473-8000 ext. 8272 352-473-1929(F) 352-745-0910(C) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make Polycom 501 go off hook when pressingany digits
Hi Mike, It's done using the digitmap feature of sip.cfg - email me offlist or come on #asterisk and I can help you with the specifics. CP On 18-Sep-06, at 11:08 AM, Mike wrote: I'm trying to make the Polycom 501 go off-hook (in speaker phone mode) when any digits is dialed and the handset hasnt been lifted. Is this possible? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to install HUDLite Server
I concur: HUDLite - couple of days, unanswered forum postings, never got it working FOP - few minutes, worked right away YMMV, CP On Sep 14, 2006, at 7:45 AM, Brodie Macleod wrote: Yeah there are some problems with the docs, and the product itself isn't very impressive -- still bugs that existed for months that basically make it worthless for me to use. Anyway, since they didn't include ircd and the perl mods in the new package, just download and install ircd-hybrid from ircd-hybrid.com, and the perl modules it references using CPAN. If you use queues in your setup, don't even bother..it still won't track calls that come in on a queue. -Brodie On Thursday 14 September 2006 12:48 am, Zeeshan Zakaria wrote: The Linux documentation on installing HUDLite doesn't really say how to install it. I can download the hudlite RPM, but where are the rest of the RPMs indicated in the documentation. And then how and where is the fonality folder is created? Somebody needs to re-write the documentaiton page. Please guide me on how to install HUD Server, if anybody has installed it successfully. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Maximum retries exceeded
This looks like a networking issue - asterisk isn't receiving any replies to signaling packets and assumes that the UA is no longer reachable. CP On 8-Sep-06, at 10:33 AM, Noc Phibee wrote: anyone know this error ?? Noc Phibee a écrit : Hi today, i have a big problems with my asterisk ... when i want call i have this error : Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Request) Sep 8 12:38:07 WARNING[28369]: chan_sip.c:1243 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. srv1*CLI for all phone and i don't have change my configuration anyone have a idea of the problems ? Thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
Or better yet, set dialplan.impossibleMatchHandling to 2. This should disable earlydial altogether. CP On Sep 8, 2006, at 2:49 PM, Eric ManxPower Wieling wrote: Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfer 3/4 digits
With respect, the problem is with your numbering plan.. CP On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote: I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get a call on 615 (snom) and transfer to 6014 it works, since snom requires me to hit ok If I get a call on 601 and transfer to 6014, than 601 will get the busy signal and I hang up as usually with transfer. Howerver the caller get the announcements: I could not get that, What could be the problem ? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent solution w/o id/password
Here's what we do:
[agent-login]
exten = s,1,NoOp(${AgentUser})
exten =
s,2,AddQueueMember(${AgentContext}|${AgentChannel}|${AgentPenalty})
exten = s,3,Wait(1)
exten = s,4,Playback(agent-loginok)
exten = s,5,Hangup
exten = s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
exten = s,104,Wait(1)
exten = s,105,Playback(agent-loggedoff)
exten = s,106,Hangup
[tax-line]
exten = s,1,Macro(dnv-messagebox-setup)
exten = s,n,Set(AgentContext=${CONTEXT})
exten = s,n,Set(AgentChannel=${CHANNEL})
exten = s,n,Set(AgentChannel=${CUT(AgentChannel,-,-2)})
exten = s,n,Set(AgentUser=${CUT(AgentChannel,/,2)})
exten = s,n,NoOp(${AgentUser})
; tax-queue agents
exten = s,n,GotoIf($[${AgentUser} = 2488-tessmanl]?:macdonap)
exten = s,n,Set(AgentPenalty=1)
exten = s,n,Goto(agent-login,s,1)
exten = s,n(macdonap),GotoIf($[${AgentUser} =
2488-macdonap]?:chengb)
exten = s,n,Goto(agent-login,s,1)
exten = s,n(chengb),GotoIf($[${AgentUser} = 2488-chengb]?:listhael)
exten = s,n,Set(AgentPenalty=2)
exten = s,n,Goto(agent-login,s,1)
exten = s,n(listhael),GotoIf($[${AgentUser} =
2488-listhael]?:nguyent)
exten = s,n,Set(AgentPenalty=3)
exten = s,n,Goto(agent-login,s,1)
exten = s,n(nguyent),GotoIf($[${AgentUser} =
2488-nguyent]?:NonAgentStart)
exten = s,n,Set(AgentPenalty=4)
exten = s,n,Goto(agent-login,s,1)
exten = s,n(NonAgentStart),BackGround(call-processors/2488)
Hope this helps.
CP
On Aug 30, 2006, at 8:55 AM, Artifex Maximus wrote:
Hello,
I'm looking for an agent managing dialplan/software/agi/whatever that
independent from asterisk queue management. I already tried this
http://www.voip-info.org/wiki/view/Agents+without+agent+channel
with no success but a lot of warning. I'm using asterisk 1.2.10 and
the dialplan above made for 1.0 might that cause the trouble.
So I'm looking for an agent management that not need agents.conf like
id and password for login. Instead if someone dial an extension from
his phone that agent (extension actually) login. If dial an another
extension he logout. If a logged in agent don't answer for a duration
automatically logoff. If no free agent on incoming call just play a
sound and hangup. This time I don't need queues just 'plain' agents
whos dynamically login/logout.
For example:
I dial 8301 and I log in with my phone (Zap, SIP, whatever). If I dial
8302 then I log off. If I don't answer for an incoming within 15 secs
asterisk automatically log me out.
If asterisk's queue managent can do this by default that would be much
better but as I see that only know the id/password solution.
bye,
Zsolt
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Re: [asterisk-users] Asterisk speaks Russian!
Westany speaks biz CP On Aug 30, 2006, at 9:50 AM, Stuart wrote: Westany, the Asterisk voice experts, announce their first Russian voice for ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Run As User Asterisk
There is a good page on the wiki about this: http://www.voip-info.org/wiki-Asterisk+non-root CP On Aug 14, 2006, at 6:44 PM, Forrest Beck wrote: Does anyone have a listing on file/directories that asterisk needs ownership of to run as a user other than root? I know about the major items --- /etc/asterisk, /var/spool/asterisk/, /var/lib/asterisk, etc... Anyone have a script to fix all the directories? Thanks in advance. FB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speed dials on Polycom IP601?
Empty line keys will be filled with speed dial entries in the phone's directory - when creating a directory entry, set the speed dial value to 1 for the first, 2 for the next.. etc. CP On 16-Aug-06, at 11:23 AM, Warren ((mailing lists)) wrote: I just got my first IP601 and put together my first * system (yay!) I have the first 2 buttons set up to be for the extension for the phone. I was wondering how I could make the remaining 4 into speed dials? IE: label button 3 Sales mgr and have it dial extension 246. TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Digium cards and HP DL servers
Hi Angel, We have two DL360s with a TE410P in each one - we had to disable USB to get the PCI slot to have an IRQ to itself. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Aug 2, 2006, at 6:38 PM, Angel Gomez wrote: Hi all. Thank's in advance. This mail is just to ask if someone can confirm that the digium/sangoma E1/T1 cards are working in the PCI-X slots of the HP DL Servers. Regards. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects
Yup - burned us a few times, too - on IP501s as well. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Aug 3, 2006, at 6:42 AM, Bill Gibbs wrote: I thought I was the only one!!! I actually replaced a phone acting just like you stated until I realized it was required the extra push as well... Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Turner Sent: Thursday, August 03, 2006 12:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] [OT] FYI: Polycom phone intermittent disconnects Just a note for Polycom phone users, that will hopefully help someone. Ever since deploying an office full of Polycom 601 phones, some users have experienced intermittent disconnects, where voice transmit dies, or both receive and transmit dies. Absolutely nothing in the Asterisk logs. Solution: plug the socket into the handset in properly! Pushing the socket in, it make a nice 'click' and _seems_ to be in, but it's not (and is a bit wobbly). Push it further, until the plastic hook is not exposed at all, and it makes another click. Now it's in :) --Jeff ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Error when calling
This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: Dear All, I have a strange problem in recieving calls on the pri the zaptel is green and everything seems very well, but when a call comes I can see the call along with the caller ID but then I get this strange message which make the call hungup: error msg: 'zap-in' from '0109687348' does not exist. Rejecting call on channel 0/18, span 1. the PRI is an E1 and I have the following configuration for extensions.conf [zap-in] exten = s,1,Answer exten = s,2,Dial(sip/100) exten = s,3,Hungup as for the zapata.conf it is as follow: [channels] language=en switchtype=euroisdn signalling=pri_cpe context=zap-in group=0 channel=1-15,17-31 I don't know what the problem is or where to look, I will appreciate it if someone can help me out? Thx MAG -- Thx MAG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ACD Queues Agents logout
Hi Kai,
This is what we do:
[agent-login]
exten = s,1,NoOp(${AgentUser})
exten =
s,2,AddQueueMember(${AgentContext}|${AgentChannel}|${AgentPenalty})
exten = s,3,Wait(1)
exten = s,4,Playback(agent-loginok)
exten = s,5,Hangup
exten = s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
exten = s,104,Wait(1)
exten = s,105,Playback(agent-loggedoff)
exten = s,106,Hangup
A.
On Jul 20, 2006, at 6:26 AM, Kai Ober wrote:
Okay, I think i have missed something:
When i use AgentCallbackLogin*(||*007) the agent is logged in, fine.
But how do i log OUT.
okay there is a timout,
autologoff=time
but how can an agent explicit log off?
regards
Kai
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Re: [asterisk-users] Caller ID on Transfers
The 'o' option to the Dial() command, along with using blind transfers, fixed this problem for us. A. On Jul 25, 2006, at 11:25 AM, Douglas Garstang wrote: I have three phones here with extensions 2944093, 3254103 and 9220371. 2944093 calls 3254103. 3254103 transfers 2944093 to 9220371. We want the caller id of 2944093 to be presented on the display of 9220371. However, the caller id of the transferer, 3254103, is appearing. This doesn't make any sense. How can we do this? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Email notification of voicemail
Aha - get rid of the leading comma for each entry.. = ,Front Desk = .. A. On Jul 13, 2006, at 1:00 PM, Kevin Savoy wrote: I've X'd out the extensions and passwords but this is all I have in there. Thanks [default] =,,Front Desk,, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Email notification of voicemail
Try having nothing after the name in your voicemail.conf: 1234 = 1234,The Marquis de Sade Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jul 12, 2006, at 11:17 AM, Kevin Savoy wrote: I have attach=no in my voicemail.conf so that can't be doing it. Not sure where that sendmail command is. Don't see it in voicemail.conf or any other config in the asterisk directory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Street Sent: Wednesday, July 12, 2006 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Kevin Savoy wrote: Asterisk is trying to send an email to users when they receive a voicemail. Can this be shut off? I have not entered any email addresses in voicemail.conf so it tries to send to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]. This of course gets rejected since the user does not exist and the root users mailbox on linux gets full of these rejection notices. I can't seem to find anywhere to tell Asterisk to stop notifying people they have voicemails. I'm using 1.2.9.1 of Asterisk. Thanks _ **Kevin Savoy** **Business Unit Telecom Analyst** 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc --- - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You could try commenting out: attach=yes Also, if you don't want any emails sent ever for any voice mail users you could probably uncomment the following line and give it a bogus path to the mailer. ;mailcmd=/usr/sbin/sendmail -t There is probably a better way to do this but we have never needed to turn it off so I am not sure. Hope this helps. -- VoIP Street Origination/Termination with SUPERIOR customer service! http://www.VoIPstreet.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Email notification of voicemail
Can you send me (or pastebin) your voicemail.conf? A. On Jul 13, 2006, at 12:35 PM, Kevin Savoy wrote: Thanks for replying. Have tried that. If I don't specify an email address it then takes the first name and last name and then the domain of the pbx. For example 1234 = 1234,Bob Smith I then get: [EMAIL PROTECTED] Which of course fails because that address doesn't exist. Any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Thursday, July 13, 2006 2:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Try having nothing after the name in your voicemail.conf: 1234 = 1234,The Marquis de Sade ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
We have just come through our busy tax season for our tax line queue on 1.2.1 with zero problems :-) Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jul 13, 2006, at 12:41 PM, Rich Adamson wrote: Warren (mailing lists) wrote: So let's cut to the chase here... If you want to run a production server with queues, which version should you be running to get 30+ days of uptime without needed a reset? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY
Yes - and it seems to prevent presence hints from working until the phone is rebooted.. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jun 26, 2006, at 9:28 AM, Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address uri=sip:[EMAIL PROTECTED];user=ip priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Out of Office Auto Reply:
Actually, if his MTA is configured properly, it shouldn't happen at all. A. On Jun 22, 2006, at 9:32 AM, Doug Geary wrote: Should only happen once if his email system is config'd in a standard method. Otherwise just *plonk* his address. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, June 22, 2006 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Out of Office Auto Reply: You got to be freaking kidding, a month of this? Cant we get an easy process for the list owner to take care of these? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quality monitoring
Care to share your Nagios plugin? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jun 22, 2006, at 9:53 AM, Curt Shaffer wrote: Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don’t really know where to start on measuring jitter other than with ICMP pulls and really don’t know where to start with doing MOS. Any ideas? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Out of Office Auto Reply:
We use MS Exchange too and, as far as I am aware, it is cognizant of mailing list headers and doesn't send OOO notices to mailing list postings. The only mailing list from which I receive my own OOO notices is one that doesn't have the proper mailing list headers set. When you receive a lot of email from outside your organization from people who expect a response, it is helpful to us (and them) if they receive OOO notifications. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jun 22, 2006, at 10:12 AM, Colin Anderson wrote: He's probably using Exchange which has a global setting to either send OOO replies to SMTP addresses or not. It's a dumbass Exchange administrator who enables this option (it is actually on by default) snip -Original Message- From: Anthony Rodgers [mailto:[EMAIL PROTECTED] Sent: Thursday, June 22, 2006 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Out of Office Auto Reply: Actually, if his MTA is configured properly, it shouldn't happen at all. A. On Jun 22, 2006, at 9:32 AM, Doug Geary wrote: Should only happen once if his email system is config'd in a standard method. Otherwise just *plonk* his address. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, June 22, 2006 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Out of Office Auto Reply: You got to be freaking kidding, a month of this? Cant we get an easy process for the list owner to take care of these? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quality monitoring
Great - thanks, Curt! A. On Jun 22, 2006, at 11:30 AM, Curt Shaffer wrote: It is really just a play on the check_icmp plugin. You could accomplish the same thing by doing the following: $USER1$/check_icmp -H $HOSTADDRESS$ -w 80.0,80% -c 100.0,100% -n 1 Where in this example it is an RTA of 80ms or 80% packet loss for a warning and 100ms or 100% packet loss for critical. The perfdata is then passed to perfparse for graphing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Thursday, June 22, 2006 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Quality monitoring Care to share your Nagios plugin? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jun 22, 2006, at 9:53 AM, Curt Shaffer wrote: Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don’t really know where to start on measuring jitter other than with ICMP pulls and really don’t know where to start with doing MOS. Any ideas? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Converting Voicemail wav to mp3
Hi Philippe, Blackberries can't play sound file attachments - wish they could. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jun 1, 2006, at 2:33 PM, Philippe Lindheimer wrote: Aaron, any chance you've gotten that mp3 email file such that a blackberry unit can listen to it? (I've experimented but the blackberry just doesn't like mp3 attachments, just links?) thanks, philippe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Memory-leak 1.2.7.1
Is there any chance you're connecting to a remote share using CIFS? What does slabtop look like on your machines? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 29-May-06, at 8:35 AM, Attilla de Groot wrote: Vij wrote: May be updatedb or some other such heavy application, which runs at night is causing heavy load on the system and spoils the working of asterisk. See if this phenomenon happens at the same time of the day everyday. Also, see what processes run at *that time*. Cheers, Vij Hi Vij, Well since the problem occurs on diffrent machines, I'm not so sure about this. I'm going to try if I can see what processes run at *that time*, but like I said it often occurs at night when I'm at sleep. So I'm first going to downgrade 1.2.3, someone told me, that he was 100% sure there are no memory leaks in that version. Greetings, Attilla ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 301's drop last two digits of dialed number
Hi Jamie, Take a look at the dialstring in your sip.cfg - you'll need to adjust this to match your local dialing plan. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 26-May-06, at 2:49 AM, Jamie Heckford wrote: Hi All, Having a rather annoying problem with the Polycom 301 phones, suspect it to be my dialplan. Basically if you lift the receiver off the handset and dial a number, it will not let you dial a number longer than 10 digits (Can see this being acceptable in US, but in UK its a right pain). As soon as the 10th digit is entered, it starts to dial and the number is invalid. If the phone is left on hook and the number is dialed, it works fine when pressing the 'send' key on the handset as it sends the whole number. Can anyone shed any light on this issue? I thought it could be asterisk is trying to Dial to soon so I added a Wait in the dialplan but it didn't seem to work. Kind regards Jamie Heckford Technical Consultant ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] US telco lingo
That would be we 48, no? :-) I think this thread needs an AK-47 now... A. On 24-May-06, at 12:33 PM, Paul wrote: If I had 47 siblings it could also mean us 48 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vancouver Asterisk Users Group
Greetings, I am trying to gauge the level of interest in an Asterisk users' group in Vancouver, BC (or in BC in general). If you would be interested, please reply off-list. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: Re: Voicemail error
Or use the newer syntax for Voicemail: exten = s,n,Voicemail([EMAIL PROTECTED]|su) Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 7-May-06, at 11:07 AM, Ira wrote: At 04:33 PM 5/6/2006, you wrote: All I need is a way to uppercase a string, which from everything I've read so far isn't in the code. Then again, I could just use all uppercase for my SIP/IAX device names even if it *does* look ugly. ;) What if you just prefix all names with the number one? 1dave 1dave-cell Ira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Message on Hold
Done with timeout=600 and queue-thankyou=path/to/sound/file in queues.conf Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 8-May-06, at 10:27 AM, Matt wrote: Hi, I know that I can have an AutoAttendent menu play when someone is in a queue to say something like Press 1 now to leave a message, or to continue holding stay on the line... However, is there anyway to prevent that from happening until the caller has been on hold for say 5 minutes? In other words, I don't want the caller to leave a voicemail UNTIL they have been on hold for 5+ minutes. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tool for Polycom configurations
Hi Bruce, We create a CSV file of our phone setup and then use shell scripts to parse them and generate mac-address.cfg, phone.cfg, sip.conf, voicemail.conf and entensions.conf entries. Contact me off list if you would like a copy now (they're not quite ready for prime-time yet) - the rest of you will have to wait until they're finished :-) but I do intend to release a bunch of monkey-level helpdesk scripts that I am working on in the near future for managing basic MAC requests. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On May 4, 2006, at 11:45 AM, Bruce Reeves wrote: I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files? -- Bruce Nortex Networks___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI got event: HDLC Bad FCS (8) on PrimaryD-channel of span
Looks like a timing problem - zaptel.conf and zapata.conf, please. A. On Apr 25, 2006, at 3:05 AM, Nico Giefing wrote: Hello, I get an Error every minute on the second card of two installed TE410P Cards in our System. The error is: PRI got event. HDLC Abort (6) on Primary D-channel of span 5(-8) PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 5(-8) Is it possible that there are known problems with 2 cards in one system? I'm running Asterisk/Libpri/zaptel from SVN branch-1.2-16008 I was running Debian Stable with Kernel 2.4.25 Since Yesterday i'm running Kernel 2.6.8 The Interrupte of the cards are: 16 and 28 Do anybody have any idea how i can solve this Problem? -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID/variable setting.
Hi Ken,
Here is what we do, if it helps:
For incoming calls (we use some Centrex lines, but want to make them
look like 4-digit locals):
; If it looks like one of ours, only show the last 4 digits
exten = s,40,GotoIf($[${CALLERIDNUM:0:8} = 60498131]?50:)
exten = s,50,SetCallerID(${CALLERIDNAME} ${CALLERIDNUM:-4})
For outgoing calls:
; If it looks like the local has already been expanded to NANPA, skip
to dialing
exten = s,3,GotoIf($[${ARG1:0:3} = 604]?20:)
; AllStream DIDs get a 604998 prefix, the rest get 604990
exten = s,4,GotoIf($[${ARG1:0:3} = 303]?:10)
exten = s,5,SetCallerID(${CALLERIDNAME} 604998${CALLERIDNUM})
exten = s,6,Goto(20)
exten = s,10,SetCallerID(${CALLERIDNAME} 604990${CALLERIDNUM})
exten = s,11,Goto(20)
Hope this helps - let me know if you need more details.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 24-Apr-06, at 9:57 AM, Ken D'Ambrosio wrote:
Hey, all. I'm trying to set my CID such that, internally, I see a
four-digit extension (which is also handy when checking VM), but
externally, I see the full 10-digit number. So I plugged these
lines into
my extensions.conf:
exten = _XXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2)
exten = _XXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1})
exten = _XXX,3,NoOp(${CALLERIDNUM})
exten = _XXX,4,Dial(${OUTBOUNDTRUNK}/${EXTEN})
(I wanted to test against my own extension, 1625; if that worked, I
wanted to strip off the 1, and then prepend the 603-123-4 to my
remaining three digits.)
Which is all well and good -- until I actually try to use it.
Then, I get:
-- Executing GotoIf(SIP/1625-f89a, 0?4:2) in new stack
-- Goto (internal,7654321,2)
-- Executing Set(SIP/1625-f89a, CALLERIDNUM=6031234625) in
new stack
-- Executing NoOp(SIP/1625-f89a, 1625) in new stack
-- Executing Dial(SIP/1625-f89a, Zap/g1/7654321) in new stack
Why does my NoOp line show my 1625 extension, when CALLERIDNUM is
-- as
far as I can tell -- being set to 6031234625? (I looked against
the Set
command page on the Wiki, and I think I'm doing it right.)
Asterisk 1.2.3, if that matters.
Thanks,
-Ken
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Re: [Asterisk-Users] still some moh troubles
Hi Bart, If it's anything like the problem we had, you are probably getting what sounds like screeching noises during MOH playback? We had this problem and made it go away by turning off hyperthreading in the server BIOS and starting Linux with noht - this was on a dual Xeon machine. Hope this helps. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Apr 20, 2006, at 6:37 AM, Bart van Daal wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: donderdag 20 april 2006 14:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] still some moh troubles Bart van Daal wrote: Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to Actually, you'll want to use ulaw for Native MOH. CUT #!/bin/sh for filename in *mp3 do eval filename=`echo $filename | cut -f1 -d.` echo Converting $filename sox -V $filename.mp3 -t au -r 8000 -U -b -c 1 $filename.ulaw resample -ql done CUT Doug Thanks for you suggestion Doug, I've converted the files using your script to ulaw but experience the same problem. A thing I forgot to mention is that it only happens on calls passing the trunks to the cisco-routers that terminate to pstn so not on internal sip-sip calls. Normal voice communication runs smoothly over the trunks it's only the moh that causes some problems. again, any pointers like those of Doug are very much appreciated thanks! Bart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Red Hat AS 4?
Hi Domenico, We're using RHEL 4 ES with no obvious issues Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Apr 21, 2006, at 3:59 AM, Mimmus wrote: Hi, I'm planning to install a new Asterisk server with a Digium TE410P card. Can I use Red Hat Advanced Server 4 (latest update)? Is this a good choice? Is recompiling Asterisk simple with kernel 2.6? Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Very high size-32 usage
Hi there, Has anyone noticed very high size-32 allocations in Asterisk servers with Digium hardware installed? Here is output from /proc/slabinfo: size-32 23763586 23763586 32 1191 : tunables 120 600 : slabdata 199694 199694 0 Here is the summary and first few rows from slabtop: Active / Total Objects (% used): 23850372 / 23890412 (99.8%) Active / Total Slabs (% used) : 204139 / 204139 (100.0%) Active / Total Caches (% used) : 95 / 134 (70.9%) Active / Total Size (% used) : 756630.62K / 760089.77K (99.5%) Minimum / Average / Maximum Object : 0.01K / 0.03K / 128.00K OBJS ACTIVE USE OBJ SIZE SLABS OBJ/SLAB CACHE SIZE NAME 23764300 23764241 -80%0.03K 199700 119798800K size-32 5085 5085 100%0.68K 10175 4068K ext3_inode_cache 51075 20557 40%0.05K681 75 2724K buffer_head 8008 3666 45%0.27K572 14 2288K radix_tree_node 9936 9863 99%0.16K432 23 1728K dentry_cache 8463 8463 100%0.12K273 31 1092K size-128 256256 100%3.00K1282 1024K biovec-(256) As you can see, almost 800MB of memory on this box is taken up with size-32 pages. This particular server is a single CPU box running Asterisk 1.2.5 and Zaptel 1.2.4 on RHEL4 and is a low-use, test box. Our two production boxes are dual 3.4GHz Xeons running Asterisk 1.2.1 and Zaptel 1.2.1 on RHEL4 SMP and exhibit the same issue (it was running into oom-killer problems with low LOWMEM on one of them that triggered all of this). Interestingly, we have an identical server to our test server that does not have Asterisk or Zaptel installed, and it does not display this issue. Has anyone else encountered this issue? What does your slabtop look like? Any thoughts or ideas would be appreciated. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 resource full problems ...
You can change the storage method on the Polycom phones from using NVRAM to VRAM to increase the number of entries (limited to 25 with NVRAM according to the Polycom Admin Guide) that a phone can store. The relevant setting is dir.local.volatile.2meg=1 or dir.local.volatile.4meg=1, depending on the model of phone you have. You then need to set dir.local.volatile.maxSize to a value between 1 and 100 to set the limit of the VRAM directory. Don't forget this note from the manual: When the volatile storage option is enabled, ensure that a properly configured boot server that allows uploads is available to store a back-up copy of the directory or its contents will be lost when the phone reboots or loses power. Hope this helps. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 14-Apr-06, at 9:16 AM, Michael Welter wrote: My customers are reporting that the contact directory can only hold about 45+ entries. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Will VoIP ITSP's be Next?
Does anyone enjoy these? It's funny - I see people being flamed for asking Asterisk questions, but not a murmur about this stuff... On Apr 13, 2006, at 5:26 PM, Bob's Leaky News Service wrote: Will VoIP be Next? snip verbal diarrhoea ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hinting
Hi Aaron, You need to create an entry in the directory of the _watching_ phone with the extension of the _watched_ phone as its contact. Set the 'Buddy watch' of this entry to 'Yes', so it appears in the list of 'Buddies' (couldn't they come up with another term for this? :-) Then, in extensions.conf, set a hint for the _watched_ extension like this: exten = 2348,hint,SIP/2348 Let me know if you have any more questions. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 3-Apr-06, at 11:42 AM, Aaron Daniel wrote: The polycoms have a buddy feature where you can watch a buddy. From what I can tell, it sends a subscribe to the server, and only works if you're hinting the phone. That's what was suggested I do since I want to be able to tell if someone's on the phone, and I've watched the sip debug as it boots up and it does in fact send a subscribe to the server for the extensions I want to watch. The server's not really doing anything with it though, so I'm kinda lost on how this is going to work. Sip debug doesn't show asterisk sending any information to the phone after it subscribes. Aaron On Mon, 3 Apr 2006, Kevin P. Fleming wrote: Aaron Daniel wrote: Ok, with the buddies, what device do you hint to? The last line of the phone? I don't understand the question... the 'buddy' is effectively a speed-dial, the same thing you would dial to call that person/ extension. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hinting
Interesting - we didn't find this on either the 501s or the 601s A. On 3-Apr-06, at 1:11 PM, Darrick Hartman wrote: Additionally, (at least on the Polycom 600's) you need to reboot your phone for this to take effect. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users
I tried to get a government/enterprise SIG or UG off the ground a number of months ago, with very limited success. If there is sufficient interest now, I could be persuaded to try again. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Mar 24, 2006, at 10:01 AM, Bob McDowell wrote: The only reason I recommended that was to protect the privacy of those on that list. I personally do not want a bunch of cold calls from asterisk 'dealers' just because I chose to implement that product. Such a list of users would make a tempting target for marketing uses... But either way, a list would be a great addition. It would go a long way toward debunking the FUD that usually accompanies a product of this type. And with Asterisk it's worse because it gets Linux FUD as well as VoIP FUD. Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Friday, March 24, 2006 11:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: FW: [Asterisk-Users] Asterisk Users Perhaps a page on the wiki would work? We could set the ground rules similar to other industries: no names, nothing more defining than a region, the number of units, etc. Would that be useful? For example, I can describe this organization as a security company in Southwest Missouri using asterisk with 60 sets and 16 lines. When you strip off my name and email, it gets a little less certain who I am talking about... Bob McDowell I like the idea of having the information on the wiki, makes it simpler for everyone to see just how well the project is doing. I'm not sure about the removing identifying information part is such a good idea, since the best way for people to trust a system is to talk to people that have used it before. Or do we just want the information to filter through the asterisk-users list? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel Meridian Opt 81C/11c and PRI
Hi Steve, Here is one of the cards that is working for us - note the downloadable D Channel card - it is very important. Your product numbers may be different in England if you're using an E1 versus our T1 in Canada. 1. ( 1 ) NTAK09BA --- 1.5 MB DTI / PRI 2. ( 1 ) NTBK51BA --- Downloadable D-Channel Handler C Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Mar 22, 2006, at 12:48 PM, Steve Rawlings wrote: I've followed the post below and have just acquired a second-user Option 11c system (rls 23.47 in the UK) now sitting on our testbench. I've tried all combinations from various posts to get this to work with our Digium TE405P but no luck. I suspect it's our PRI in the Option 11, it's an NTAK79. It's been suggested I need an NTBK50 instead. Can anyone confirm, which PRI are successful 11's using? Thanks. Steve - Original Message - From: Greg Camp [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, January 24, 2006 3:07 PM Subject: [Asterisk-Users] Nortel Meridian Opt 81C and PRI We've been trying unsuccessfully to connect our Meridian Option 81C to a TE110P via PRI. We've followed the directions in asterisk-meridian-a1.pdf (link on http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it doesn't seem to work on our 81C (even though many, many users report it works very well on Option 11's). Has anyone had any success in getting the above combination to work with Asterisk? The results we get seem to vary depending on how closely we follow the reference guide that Andy put together. If we follow it exactly, the d-channel comes up, but the b-channels stay in MBSY instead of IDLE. I can post more details and config files if requested, but I'm curious if anyone has successfully made an Option 81C work with PRI to an Asterisk box. Thanks, Greg [EMAIL PROTECTED] Excell Services 806-747-2474 806-747-5047 fax ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI DMS100 - Nortel Meridian Option 81
Hi Greg, I'll dig it out - we only expand the outgoing callerID to 10 digits for external (PSTN) calls, so we don't have the CID issues you mention. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Mar 23, 2006, at 6:21 AM, Greg Camp wrote: Anthony, We had tried using 5ESS, but instead of seeing 4-digit extensions on the Asterisk box we would see the entire 10-digit caller-id value (I assume because Nortel sees it as an external T1). I will try a setup using NI2 on both sides. But if you could provide some more specifics (both for Asterisk and Nortel) it would be greatly appreciated. Thanks, Greg -Original Message- From: Anthony Rodgers [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 22, 2006 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PRI DMS100 - Nortel Meridian Option 81 Hi Greg, Our experience is that both Asterisk and Nortel are capable of understanding DMS100 enough to each be able to connect to a real DMS100 - however neither is capable of actually being a DMS100. We actually ended up using 2 PRIs between our Nortel 11C and Asterisk - the first is set up as a tie trunk in the Nortel and uses NI2 on the Asterisk side. This setup allows us to receive caller ID information from the Nortel and is used only for calls from the Nortel to Asterisk. The second PRI is set up as a 5ESS trunk so that the Nortel will accept caller ID from Asterisk and is used only for calls from Asterisk to the Nortel. If you need more specific details, let me know. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Mar 22, 2006, at 3:21 PM, Greg Camp wrote: Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100 resetinterval=72000 signalling=pri_net channel=1-23 The Asterisk box will see the call setup message, but according to the d-channel trace (below) a RELEASE(77) message happens shortly after the CALL PROCEEDING(2) message. The effect is that calls between the two systems do not happen. Can someone versed in d-channel messages determine what is going on here? Also, is there any way to tell the Zaptel card to emulate a particular release version for DMS100? I believe the Meridian is expecting Release 36, or something like that (we've tried leaving Release ID blank on the Meridian side with the same results). Protocol Discriminator: Q.931 (8) len=42 Call Ref: len= 1 (reference 20/0x14) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 04 e9 80 83 14] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 DS1 Identifier: 0 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 0 Channel: 20 ] [28 0a b1 47 52 45 47 20 43 41 4d 50] Display (len=10) Charset: 31 [ GREG CAMP ] [6c 06 09 80 34 32 32 34] Calling Number (len= 8) [ Ext: 0 TON: Unknown Number Type (0) NPI: Private Numbering Plan (9) Presentation: Presentation permitted, user number not screened (0) '4224' ] [70 05 e9 34 39 39 31] Called Number (len= 7) [ Ext: 1 TON: Abbreviated number (6) NPI: Private Numbering Plan (9) '4991' ] -- Making new call for cr 20 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 40 (cs0, Display) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 20/0x14) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 94] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 20 ] -- Accepting call from '4224' to '4991' on channel 0/20, span 1 Protocol Discriminator: Q.931 (8
Re: [Asterisk-Users] PRI DMS100 - Nortel Meridian Option 81
Hi Greg, Our experience is that both Asterisk and Nortel are capable of understanding DMS100 enough to each be able to connect to a real DMS100 - however neither is capable of actually being a DMS100. We actually ended up using 2 PRIs between our Nortel 11C and Asterisk - the first is set up as a tie trunk in the Nortel and uses NI2 on the Asterisk side. This setup allows us to receive caller ID information from the Nortel and is used only for calls from the Nortel to Asterisk. The second PRI is set up as a 5ESS trunk so that the Nortel will accept caller ID from Asterisk and is used only for calls from Asterisk to the Nortel. If you need more specific details, let me know. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Mar 22, 2006, at 3:21 PM, Greg Camp wrote: Hello all, I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option 81C system. The PRI line is currently setup as DMS100. Here are the relevant lines from zaptel.conf and zapata.conf: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone = us zapata.conf: [channels] language=en context=from-internal musiconhold=default switchtype=dms100 resetinterval=72000 signalling=pri_net channel=1-23 The Asterisk box will see the call setup message, but according to the d-channel trace (below) a RELEASE(77) message happens shortly after the CALL PROCEEDING(2) message. The effect is that calls between the two systems do not happen. Can someone versed in d-channel messages determine what is going on here? Also, is there any way to tell the Zaptel card to emulate a particular release version for DMS100? I believe the Meridian is expecting Release 36, or something like that (we've tried leaving Release ID blank on the Meridian side with the same results). Protocol Discriminator: Q.931 (8) len=42 Call Ref: len= 1 (reference 20/0x14) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 04 e9 80 83 14] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 DS1 Identifier: 0 Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 0 Channel: 20 ] [28 0a b1 47 52 45 47 20 43 41 4d 50] Display (len=10) Charset: 31 [ GREG CAMP ] [6c 06 09 80 34 32 32 34] Calling Number (len= 8) [ Ext: 0 TON: Unknown Number Type (0) NPI: Private Numbering Plan (9) Presentation: Presentation permitted, user number not screened (0) '4224' ] [70 05 e9 34 39 39 31] Called Number (len= 7) [ Ext: 1 TON: Abbreviated number (6) NPI: Private Numbering Plan (9) '4991' ] -- Making new call for cr 20 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 40 (cs0, Display) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 20/0x14) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 94] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 20 ] -- Accepting call from '4224' to '4991' on channel 0/20, span 1 Protocol Discriminator: Q.931 (8) len=8 Call Ref: len= 1 (reference 20/0x14) (Originator) Message type: RELEASE (77) [08 02 81 e4] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/20, span 1 got hangup -- Executing Macro(Zap/20-1, exten-vm|novm|4991) in new stack -- Executing Macro(Zap/20-1, user-callerid) in new stack -- Executing DBget(Zap/20-1, AMPUSER=DEVICE/4224/user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=4224/user -- DBget: Value not found in database. -- Executing Macro(Zap/20-1, hangupcall) in new stack -- Executing ResetCDR(Zap/20-1, w) in new stack -- Executing NoCDR(Zap/20-1, ) in new stack -- Executing Wait(Zap/20-1, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'Zap/20-1' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'Zap/20-1' NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q
Re: [Asterisk-Users] Problem compiling zaptel on latest RHEL kernel(2.6.9-34.EL)
Many thanks, Russ - I'll give this a try. Thank goodness a) for test servers and b) for the ability of Linux to rollback with a simple change to grub.conf :-) Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 11-Mar-06, at 7:33 AM, Russ Price wrote: Anthony Rodgers wrote: Greetings, I have just updated our test server to 2.6.9-34.EL and get the following error messages when compiling zaptel: make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zaptel/zaptel-1.2.1/zaptel.o /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: error: syntax error before zone_lock [snipped] This bit me with CentOS 4.2 as well. The problem is actually a typo in the kernel spinlock.h file. See: http://bugs.digium.com/view.php?id=6425 and https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=180568 for more information. Here's a quick fix. In your zaptel Makefile, add the following (line 38 for 1.2.4) - THIS SHOLD BE ALL ONE LINE: CFLAGS+=$(shell if uname -r | grep -q 2.6.9-34.EL; then echo -Drw_lock_t=\rwlock_t\; fi) This way, if this is fixed in the next kernel release, you won't need to make another change to the Makefile. Russ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
AFIAK, they can't - we would like to do the same thing, but it's not possible with patching the source. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 10-Mar-06, at 7:56 PM, btb wrote: can the default voicemail folders (old, work, friends, etc.) be changed? for example, i'd like to configure asterisk so that there are only folders called friends and old. thanks -ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to connect 3 or more servers via IAX ?
Hi Jean-Louis, We have 3 servers connected togther - we do it by creating specific trunks between each one. ### iax.conf from asterix server: ; IAX Trunks [dogmatix-in] type=user auth=md5 host=voip.dogmatix.dnv.org secret= context=international trunk=yes [dogmatix-out] type=peer auth=md5 host=voip.dogmatix.dnv.org username=asterix-in secret= context=international trunk=yes [obelix-in] type=user auth=md5 host=voip.obelix.dnv.org secret= context=international trunk=yes [obelix-out] type=peer auth=md5 host=voip.obelix.dnv.org username=asterix-in secret= context=international trunk=yes ### iax.conf from dogmatix server ; IAX Trunks [asterix-in] type=user auth=md5 host=voip.asterix.dnv.org secret= context=international trunk=yes [asterix-out] type=peer auth=md5 host=voip.asterix.dnv.org username=dogmatix-in secret= context=international trunk=yes The iax.conf from the obelix server would be similar. Hope this gives the idea OK - let me know if you need any more information. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 11-Mar-06, at 8:04 AM, Jean-Louis curty wrote: Hi, I successfully connected 2 servers via IAX but I'm pulling my hair to connect 2 extra servers , Anyone connected 3 or 4 servers together ? is it possible ? I d like to share the dialplan so _2 goes to server A _3 goes to serverB _4x goes to server C etc from the 4 servers any example of which one is peer, which one is user or friend would help me :-) thanks jl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Bug?
Unless I'm reading our CDR data wrong, such calls only generate one record for the actual answered call since we started way back on 1.0.9. Here's a sample record: ,6044378358,2380,ITS,6044378358,Zap/9-1,SIP/luv- c57d,Dial,SIP/luvSIP/luv-computerroombackSIP/luv-computerroomfron tSIP/luv-itsresourcec,2006-03-03 13:13:36,2006-03-03 13:13:43,2006-03-03 13:16:02,146,139,ANSWERED,DOCUMENTATION 4 UAs are dialed - only one answered the call - only one CDR record. Hope this helps. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 13-Mar-06, at 5:32 PM, Damon Estep wrote: Trying to figure out if a bug report should be submitted. Can anyone on 1.2.x verify of this has been corrected? I am on CVS 8/2005 If a call comes in to an extension that dials more than one channel (rings at more than one phone) both calls in the CDR show a status of answered when only one is answered, the source channel is bridged to only one of the two destination channels, but both CDRs show answered. It looks as if the status is taken from the source channel, not the destination channel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem compiling zaptel on latest RHEL kernel (2.6.9-34.EL)
Greetings, I have just updated our test server to 2.6.9-34.EL and get the following error messages when compiling zaptel: make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zaptel/zaptel-1.2.1/zaptel.o /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: error: syntax error before zone_lock /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: error: incompatible types in initialization /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: error: initializer element is not constant /usr/src/zaptel/zaptel-1.2.1/zaptel.c:383: warning: data definition has no type or storage class /usr/src/zaptel/zaptel-1.2.1/zaptel.c:384: error: syntax error before chan_lock /usr/src/zaptel/zaptel-1.2.1/zaptel.c:384: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel-1.2.1/zaptel.c:384: error: incompatible types in initialization /usr/src/zaptel/zaptel-1.2.1/zaptel.c:384: error: initializer element is not constant /usr/src/zaptel/zaptel-1.2.1/zaptel.c:384: warning: data definition has no type or storage class /usr/src/zaptel/zaptel-1.2.1/zaptel.c:187: warning: 'fcstab' defined but not used make[2]: *** [/usr/src/zaptel/zaptel-1.2.1/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel/zaptel-1.2.1] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686' make: *** [linux26] Error 2 If I reboot from the previous kernel 2.6.9-22.0.2.EL, zaptel compiles just fine. This behavior is true for both zaptel-1.2.1 (shown above) and zaptel-1.2.4. Thoughts? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy (S101) echo?
Hi Bradley, Yes, I experienced quite a lot of echo with my IAXy, until I switched analog handsets - in my case, it was severe acoustic coupling in a cheap handset. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Mar 7, 2006, at 11:38 AM, Bradley M. Kuhn wrote: I just purchased an IAXy (S101) for a home setup; I've become a de-facto expert on Asterisk for work. Everything is working great, but I notice a substantial echo on calls connected through the IAXy to POTS telephones. Has anyone encountered something similar and found a solution? I found some posts about this in the past few years, but never any replies. The Wiki on voip-info.org doesn't seem to have anything about it; I'd be happy to condense any replies I receive to information to put up there. Thanks! -- bkuhn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + SE Linux
Hi Yusuf, All our * boxes have SELinux installed and active - we haven't had to make any changes to the default SELinux config to make * work properly. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Mar 7, 2006, at 7:15 AM, yusuf wrote: Hi guys, I am busy planning to implement SE Linux on my asterisk box. Either that or I will use AppArmor from Suse. I just want to know what are others experiences/incidents with SE Linux or AppArmor thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with T1 installation
Are you sure you're supposed to be using EM?
On Feb 24, 2006, at 5:39 AM, Nitin Joshi wrote:
Hi All,
I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its
connected directly to the PSTN. But I am unable to make outbound calls
on the zap channels. The light on the card is green. Asterisk CLI
shows all 24 channels when I give the command 'zap show channels'. I
also noticed that Asterisk CLI shows an incoming call every few
seconds on the 24th channel. This must be some kind of a timing
signal. This is he first time I am configuring a T1 so I must have
done something wrong I guess.
These are the commands I used to load the zap module:
modprobe zaptel
modprobe wcte11xp
ztcfg -vvv
---
my zaptel.conf is as follows:
span=1,1,0,esf,b8zs
em=1-24
loadzone = us
defaultzone=us
--
the zapata.conf is as follows:
[trunkgroups]
[channels]
group=1
language=en
signalling=em_w
usecallerid=yes
callerid=asreceived
context=default
echocancel=64
echocancelwhenbridged=yes
rxgain=1.0
txgain=1.0
channel = 1-2
group=2
language=en
signalling=em_w
usecallerid=yes
callerid=asreceived
context=default
echocancel=64
echocancelwhenbridged=yes
rxgain=1.0
txgain=1.0
channel = 3-24
--
In extensions.conf i have specified the following line:
[default]
exten = _ZX,1,Dial(zap/g1/${EXTEN},15,tr)
--
When I try to dial using the T1 line I get the following error :
Feb 24 06:56:53 NOTICE[5724]: app_dial.c:1010 dial_exec_full: Unable
to create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/7180-a103' status is 'CHANUNAVAIL'
Any ideas guys?
Thanks and regards,
Nitin Joshi.
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Re: [Asterisk-Users] is there a web interface to this mailing list?
You'll likely find Asterisk itself even more of a challenge then. On Feb 15, 2006, at 1:29 PM, roswel ajf wrote: hi, To post, and to reply to a post, i have to goto my email. Now, if there is a web interface to these mailing list, things would be easier. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] odd 'digital' sound artifacts
Your output looks like you have 3 cards, two of which are sharing interrupts - or am I missing something? On Feb 10, 2006, at 7:04 AM, Gerard Saraber wrote: So nobody heard these before? or did I do something stupid that anyone should know and nobody wanted to yell at me for it ;) On Wed, 2006-02-08 at 12:54 -0600, Gerard Saraber wrote: Hi, I've got some weird sound artifacts happening during calls, they're very hard to describe, so I have a 122kb recording: http://openprojects.rarcoa.com/~miztic/artifact.wav normally the artifacts are just short blips, not quite as long as the one above, but they sound the same. When using the aggressive echo suppressor, it seems like those artifacts cause a really loud buzzing sound to come out of the cisco phone, pretty much made using the aggressive canceler impossible to use, it's too bad because it worked the best out of all of them, mark3 works ok but still gives echos on at least 20% of the calls. I thought they might be caused by IRQ sharing, so I pulled one of the TDM400P cards out and made sure the remaining two were on their own IRQ, the artifacts were still there. I've also tried running a kernel with all the low-latency stuff turned on, and the same kernel with it all turned off (2.6.16-rc2) doesn't appear to make any difference either. I'm not sure what else to try, any input would be appreciated. Thanks, Gerard Saraber [EMAIL PROTECTED] hardware: AMD64 1.8Ghz 512M ram MSI nforce3 socket 754 mainboard 3 Digium TDM400P cards, 10 FXO + 2 FXS modules /proc/interrupts CPU0 0: 2784232 IO-APIC-edge timer 1: 8 IO-APIC-edge i8042 8: 0 IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 71552 IO-APIC-level eth0 185: 9412 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217: 5577811 IO-APIC-level wctdm, wctdm 225: 2769262 IO-APIC-level wctdm lspci (for completeness): 02:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at ac00 [size=256] Memory at fdeff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 225 I/O ports at a800 [size=256] Memory at fdefe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at a400 [size=256] Memory at fdefd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel Meridian Opt 81C and PRI
On Feb 8, 2006, at 9:27 AM, Greg Camp wrote:
Now, our latest two issues:
1) When a user on the Nortel makes a call to a user on * a 10-digit
callerid value shows up on the SIP phone instead of the users
extension.
Has anyone encountered this and found a work-around? It's been
suggested that we use a QSIG interface instead of 5ESS emulation, but
did not purchased the Nortel QSIG option so it is unavailable.
We implemented a macro that stripped the leading 6 digits from the
numbers, like this:
; If it looks like one of ours, only show the last 4 digits
exten = s,40,GotoIf($[${CALLERIDNUM:0:8} = 60498131]?50:)
exten = s,50,SetCallerID,${CALLERIDNAME} ${CALLERIDNUM:-4}
2) We would like to use Comedian Mail for company wide voicemail. I
can
setup user extensions easily enough. I have also setup two 4-digit
extensions; one for picking up voicemail and one for leaving voicemail
for an arbitrary user. The second ext is used primarily by the
receptionist (coming from the Nortel PBX) to redirect callers to users
voicemails. The issue I'm having is that if you don't pass an
extension
to the Voicemail() function * will prompt you one time. If you key the
ext incorrectly the system hangs up on you. Is there a way to prompt
the caller for the extension to leave a message for, accept the ext,
check the database, and give the caller another chance if the ext is
invalid?
AFAIK, Voicemail() will jump to n+101 if the requested mailbox doesn't
exist - you can use that to return to the prompt asking for the mailbox
number.
Thanks,
Greg
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