[asterisk-users] Re: PSTN Calls from SI.P: buzzing and pops
Tzafrir Cohen wrote: On Fri, Feb 16, 2007 at 12:39:54AM -0500, Allen Casteran wrote: We have SIP phones connecting to *, and our PSTN lines connecting through an Astribank FXO. Internal Sip-SIP calls are clear. External calls through the Astribank get occassional low level buzzing for about 1/2 - 1 sec at a time. Those are usually accompanied by some light pops or static No transcoding, all ulaw. PSTN lines come from an Adtran IAD channel bank about 10 ft of cable from the Astribak. Astribank and Adtran are both chassis grounded to our UPS. Tried set echocancel=on and echocancel=off in the zapata.conf... No difference. You mentioned (in another mail) that if you connect the same line to the PSTN the line is clear. Was it in the same setup or were there some resets and/or reconfigurations in the process? If you connect a simple phone insted of the Astribank, do yu have the same problems? Same setup. We replaced analog phones with the * Server and then noticed the buzzing. Before installing the * server the analog phones had been used on the same lines with no problems for about a year. No changes have been made to the lines themselves. The buzz occurs every minute or so, but not on any schedule that I can determine. (ie not every 65 seconds). I have done several config changes on the * server to try and eliminate the problem to no avail. ___ --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] Fanless solution
On 14 Feb 2007, at 16:37, shadowym wrote: Hi there, I'm looking for a compact fanless solution preferrably wall mountable and not too exotic. It needs to be commercial grade. I don't really consider most of the Via ITX solutions I have seen commercial grade but perhaps someone can convince me otherwise. No, I don't think they are commercial grade. I have 2 dead Via ITX motherboards out of 6 I've bought in the last couple of years. 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
[asterisk-users] Pickup application
I am trying to configure the pickup.
This is my dialplan:
exten = _57.,1,Pickup(${EXTEN:2})
So, when i call for example 57333 Asterisk tries to pick up the call
ringing on 333
The problem is that it works only with internal calls!
For example, if i call 333 from 334 and while 333 i ringing i try to
dial 57333 it works.
If i call an external number that via dialplan dials 333 dialing 57333 i got:
Executing [EMAIL PROTECTED]:1] Pickup(SIP/200-08432290, 333) in new stack
[Feb 16 09:20:46] NOTICE[7586]: app_directed_pickup.c:159 pickup_exec:
No target channel found for 333.
Where am i wrong?
Thanks
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[asterisk-users] Re: PSTN Calls from SI.P: buzzing and pops
As a follow up test, I unplugged line 1 from the Astibank and connected it to a butt set. Placed a call into my own office and listened to 10 minutes of voice mail. Clean sound. No crackles or buzzing. Reconnect the CO line to the Astribank and place same call to my office. Listen to voicemail and hear 3 pops, buzzes in 2-3 minutes. Ideas? ___ --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] Sending SMS from Asterisk
On 14 Feb 2007, at 17:35, Stephen Bosch wrote: Tim Panton wrote: We've used www.Simplewire.com , they have a x86 linux executable which we wrap in a shell script and call from the dialplan with a System() call. We've been happy with them for years. Wow! Are these guys in Canada? (One of the sample numbers was a 416 area code, which is in Toronto). I tried a sample message -- it arrived in 2 seconds. That is better performance that Rogers' own web interface! More information, please! Erm, what do you want to know ? It works, they have APIs in most languages, they have been in business for years, they aren't the cheapest to Europe. Feel free to contact me off-list. 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
Re: [asterisk-users] moving WiFi phone
On 15 Feb 2007, at 09:54, Alberto Pastore wrote: Pavel Jezek ha scritto: Jens Vagelpohl wrote: I have two APs (Apple AirPorts) sending on the _same_ channel. Handover works perfect with no discernible loss of connectivity or audio using a Siemens SL75. The handover cannot even be noticed. as I know, best practice says, that neighboring AP should use _non overlapping_ channels... :-\ After *months* of troubles using a 14 APs network with same SSID, WPA/TKIP security model, tx power settings and channels carefully distributed in order to be as non overlapping as possible, including a controller capable of performing fast layer2 reauthentication (e.g. something like caching WPA keys between access points), I always got VERY POOR roaming performance. I've tested these phones: UTStarcom F1000 UTStarcom F1000g UTStarcom F3000g Siemens Gigaset SL75 WLAN Nokia E60 Nokia E70 Samsung WIP6000 Linksys WIP300 I was desperate. I took a bold step. I downgraded to WEP-128 (I know it's weak) and, despite the recommendations from any good wifi networking guide, I SET ALL APs ON THE SAME CHANNEL. Don't ask me why, but now roaming is PERFECT, never had a call dropped or even a hiss or crackling noise during conversation. I can even run or move over the site hangar on forklift trucks while talking on the phone, at 15-20 mph. Luckily there are no high throughput demands for data transmission (PDAs, notebooks, etc) over the wifi network, so I didn't got performance issues. Imho roaming support on 802.11 wifi networks is far from being usable... The WAP54's have a 'repeater' mode which I've used on occasion. 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
Re: [asterisk-users] Re: PSTN Calls from SI.P: buzzing and pops
On Fri, Feb 16, 2007 at 03:27:15AM -0500, Allen Casteran wrote: As a follow up test, I unplugged line 1 from the Astibank and connected it to a butt set. Placed a call into my own office and listened to 10 minutes of voice mail. Clean sound. No crackles or buzzing. Reconnect the CO line to the Astribank and place same call to my office. Listen to voicemail and hear 3 pops, buzzes in 2-3 minutes. uname -r cat /proc/xpp/sync -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Re: PSTN Calls from SI.P: buzzing and pops
Tzafrir Cohen wrote: On Fri, Feb 16, 2007 at 03:27:15AM -0500, Allen Casteran wrote: As a follow up test, I unplugged line 1 from the Astibank and connected it to a butt set. Placed a call into my own office and listened to 10 minutes of voice mail. Clean sound. No crackles or buzzing. Reconnect the CO line to the Astribank and place same call to my office. Listen to voicemail and hear 3 pops, buzzes in 2-3 minutes. uname -r 2.6.9-42.0.8.ELsmp cat /proc/xpp/sync # To modify sync source write into this file: # HOST- For host based sync # 0 0 - XBUS-0/XPD-0 provide sync # m n - XBUS-m/XPD-n provide sync HOST tick: #6109170 tick rate: 1000/second (SAMPLE_TICKS=1) ___ --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: PSTN Calls from SI.P: buzzing and pops
On Fri, Feb 16, 2007 at 04:00:36AM -0500, Allen Casteran wrote: Tzafrir Cohen wrote: On Fri, Feb 16, 2007 at 03:27:15AM -0500, Allen Casteran wrote: As a follow up test, I unplugged line 1 from the Astibank and connected it to a butt set. Placed a call into my own office and listened to 10 minutes of voice mail. Clean sound. No crackles or buzzing. Reconnect the CO line to the Astribank and place same call to my office. Listen to voicemail and hear 3 pops, buzzes in 2-3 minutes. uname -r 2.6.9-42.0.8.ELsmp cat /proc/xpp/sync # To modify sync source write into this file: # HOST- For host based sync # 0 0 - XBUS-0/XPD-0 provide sync # m n - XBUS-m/XPD-n provide sync HOST tick: #6109170 tick rate: 1000/second (SAMPLE_TICKS=1) Any change if you run: echo 0 0 /proc/xpp/sync (or maybe instead of 0: the number of your FXO unit). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Re: PSTN Calls from SI.P: buzzing and pops
Tzafrir Cohen wrote: Any change if you run: echo 0 0 /proc/xpp/sync (or maybe instead of 0: the number of your FXO unit). That may be it. Let's see how it works for the girls in the office in the morning. I'll send you a note before 17:00. I did try that setting earlier, but I think that was during a period that the trunks would not come up for me. As a result I may never have changed that for a proper test. Thanks for now :) ___ --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] Fanless solution
On Fri, 16 Feb 2007, Tim Panton wrote: On 14 Feb 2007, at 16:37, shadowym wrote: Hi there, I'm looking for a compact fanless solution preferrably wall mountable and not too exotic. It needs to be commercial grade. I don't really consider most of the Via ITX solutions I have seen commercial grade but perhaps someone can convince me otherwise. No, I don't think they are commercial grade. I have 2 dead Via ITX motherboards out of 6 I've bought in the last couple of years. And I have over a dozen newish Via CN1000 boards ticking away quietly in various applications (mostly asterisk), 2 EK1000 boards (with variable speed fans which never seem to come on) acting as routers, and 3 older (4 years) 533MHz boards still in daily use as my asterisk RD systems, all without issues so-far... ... which doesn't necessarily mean they are prefect, but they are working for me. The issue with commercial grade for me, at least is making the box not look like a PC - there are mini ITX mobos with headers rather than on-board sockets, etc. but then there's the additional engineering required to put them in a custom box, and I'm not quite ready for that yet! Wall mounting case: http://www.icp-epia.co.uk/index.php?act=viewProdproductId=77 but it's a bit ugly (but does anyone care with wall mounting cases? Anyone know of prettier ones I can get in the UK?) There is a fanless Commell board: http://www.icp-epia.co.uk/index.php?act=viewProdproductId=99 but it's only 600MHz, however it has 512KB of cache - the Via ones only have 128KB, so that might make a little difference - My RD systems at 533 MHz and 64KB cache seem very capable of running a small offices asterisk needs - half a dozen handsets and a TDM400 card with 2 analogue lines doesn't seem to impose any load on them at all... (GSM transcoding does, however, but I've never tested them to their limits) 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
RE: [asterisk-users] Fanless solution
Try Orbit Micro They have network appliance systems that are definitely commercial grade http://store.orbitmicro.com/ Femi ___ --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] Hint and CallerID
Carlos Chavez schrieb: On Thu, 2007-02-15 at 16:11 +0100, Tobias Wolf wrote: Callerid is not defined by the hints. You need the line: callerid=asreceived This should be in the definition of your zap channel so it passes the callerid information without modification to your phones. On the Phone which is called, the callerid shows up just fine, only the Snom Phone, which shows me an incoming call by the blinking lights, doesn't tell who is calling and what number he has dialed. That would be nice to see. Well, it seems that this cannot be accomplished by hints. Thx for your answer. Tobias ___ --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] Anyone know what this warning is about? Nothing in list history about it either..
20 jan 2007 kl. 03.01 skrev Eric Bishop: On inbound calls from my SIP provider I get multiple warnings as follows: WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid host Everything else works but these warnings are a pain and I don't know what they are about Nothing on previos lists or Google explains... ___ You have a bad host name that does not resolve in DNS in the via header of a SIP message, propably an INVITE coming in. /O ___ --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] 64 bit HPEC modules available?
I am running 64 bit linux on my Asterisk box and would like to get the new HPEC software running on it. However, while there are 32 bit modules available, there are no 64 bit modules on the ftp site: http://ftp.digium.com/pub/telephony/hpec/64-bit/ In some places on the digium website it states 32 bit only and other places including the documentation it states 32 64 bit are available. Is there a 64 bit version of the HPEC module or are the 32 bit modules suitable (can't imagine that they would be). Thanks in advance Greg ___ --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] freepbx with ASTERISK 1.4
Hi everybody, it's possible to configure freepbx 2.2 with asterisk 1.4? Have a nice day Younss AZ KASTERISK.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
Re: [asterisk-users] Re: Long call setup times on SIP to zaptel calls
Benny Amorsen wrote: EW == Eric \ManxPower\ Wieling Eric writes: EW All of our SIP phones dial instantly when the users finished EW dialing. We can do this because we have no ambiguous extension EW lengths. i.e. no _XXX and _ and we don't use the . pattern EW match. If you have managed that even for international calls, I'm impressed. The dial plan of Sweden is variable-length, so you would have to know the length of every area code. Obviously possible, but a pain to program and keep up-to-date. After I pressed send, I realized that *someone* would bring that up. 8-) My users do virtually no calling to numbers outside of the USA. The ONLY . in my dialplan is for international calls. ___ --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] End Wrap-up Time?
Indeed it does. And you could simply call Pause, Wait1, Unpause as an Wrap-Cancel application. I don't see any repercussions. What if we patched Asterisk to do just that? What could the repercussions be? They're already pausing/unpausing, so having the wrapup time auto-zero on unpause seems a non-issue... -A. ___ --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] freepbx with ASTERISK 1.4
Yes check the freepbx website, and in particular trac bug #1610. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of younss azzayani Sent: 16 February 2007 11:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] freepbx with ASTERISK 1.4 Hi everybody, it's possible to configure freepbx 2.2 with asterisk 1.4? Have a nice day Younss AZ KASTERISK.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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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
[asterisk-users] Asterisk callerID
Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I make a call directly using one of this line, the callerID is sending correctly. With the same zapata config file and the Freepbx 2.1.3, the callerId was sending correctly. Any clue will be welcome Thanks in advance. VoipCrazy -- zapata.conf-- [channels] language=en context=from-zaptel rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes callerid=asreceived hidecallerid=no callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=50 immediate=no rxgain=3.0 txgain=4.0 immediate=no busydetect=yes busycount=8 answeronpolarityswitch=yes hanguponpolarityswitch=yes ringtimeout=8000 faxdetect=both signalling=fxs_ks useincomingcalleridonzaptransfer=yes channel = 1-2 ___ --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] Re: Pickup application
Try NVPickup.
--
Date: Fri, 16 Feb 2007 09:24:08 +0100
From: nik600 [EMAIL PROTECTED]
Subject: [asterisk-users] Pickup application
I am trying to configure the pickup.
This is my dialplan:
exten = _57.,1,Pickup(${EXTEN:2})
So, when i call for example 57333 Asterisk tries to pick up the call
ringing on 333
The problem is that it works only with internal calls!
For example, if i call 333 from 334 and while 333 i ringing i try to
dial 57333 it works.
If i call an external number that via dialplan dials 333 dialing 57333 i got:
Executing [EMAIL PROTECTED]:1] Pickup(SIP/200-08432290, 333) in new stack
[Feb 16 09:20:46] NOTICE[7586]: app_directed_pickup.c:159 pickup_exec:
No target channel found for 333.
Where am i wrong?
Thanks
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Re: [asterisk-users] Re: Pickup application
On 2/16/07, Justin Newman [EMAIL PROTECTED] wrote: Try NVPickup. -- Sorry, but it seems to doesn't exists... WARNING[10208]: pbx.c:1755 pbx_extension_helper: No application 'NVPickup' for extension (from-internal, 57333, 1) ___ --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] Symbian IAX client
I also would like to know if there is an application like this. The most i've tried in a mobile device is using PPCIAX for the pocketpc. Any comments also on the feasibility of developing something like this if the application is not yet available. On 2/15/07, Peter Spikings [EMAIL PROTECTED] wrote: Hi all, Does anyone know of an IAX client for Symbian? I have an e61 and would like to make calls through my home Asterisk box from places where I have WiFi access, as NAT is in the way I suspect that it'll be a pain to get SIP working like that as the NAT router doesn't do SIP connection tracking. Thanks, Peter. This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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] Jabber/Asterisk Integration
Started playing with 1.4 and I'm curious what uses people have come up with for the Jabber integration? So far I can think of presence based call routing, but I'm sure there are other ideas. How are YOU using the new Jabber features in 1.4? :) -- Kyle Sexton ___ --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] Fanless solution
On Fri, 16 Feb 2007, Femi wrote: Try Orbit Micro They have network appliance systems that are definitely commercial grade http://store.orbitmicro.com/ Hehe... http://store.orbitmicro.com/ccp2889-compact-embedded-system-w--onboard-via-c3-1ghz-pr-ebs-1569ps-1-101920.htm ;-) 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
Re: [asterisk-users] Symbian IAX client
On 16 Feb 2007, at 12:46, Vernier Umali wrote: I also would like to know if there is an application like this. The most i've tried in a mobile device is using PPCIAX for the pocketpc. Any comments also on the feasibility of developing something like this if the application is not yet available. On phones that already support SIP it should be do-able. But on ones that don't you have to look out for the fact that on some designs you can't get bidirectional audio from the audio DSP to the application processor, the data path is one way or the other. I got burnt by that on the Savaje OS. Do you think that there is a market for this? If so contact me off-list. 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
[asterisk-users] Digium TE110P
Hi I am currently installing a TE110P. SUSE10 The zttest test result is : average 99.9991%. My server : processor Intel® Celeron® D 330, 2.66 GHz, cache 256 Ko, FSB 533 MHz , 1G RAM. Hope it can help. Now I have a question to TE110P users : The card is physically plugged, modprobe, ztcfg ok etc ... There is a red led blinking. The question is : should this led provide a green continuous light or is it correct to have a red blinking light ? Thanks in advance Olivier Envoyez vos cartes de voeux depuis www.laposte.net Elles seront ensuite distribuées par le facteur : pratique et malin ! ___ --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] Symbian IAX client
Yeah, it would be very neat. NAT is such a pain, roll on IPv6 :) On Fri, 2007-02-16 at 20:46 +0800, Vernier Umali wrote: I also would like to know if there is an application like this. The most i've tried in a mobile device is using PPCIAX for the pocketpc. Any comments also on the feasibility of developing something like this if the application is not yet available. On 2/15/07, Peter Spikings [EMAIL PROTECTED] wrote: Hi all, Does anyone know of an IAX client for Symbian? I have an e61 and would like to make calls through my home Asterisk box from places where I have WiFi access, as NAT is in the way I suspect that it'll be a pain to get SIP working like that as the NAT router doesn't do SIP connection tracking. Thanks, Peter. This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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 This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --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] Jabber/Asterisk Integration
I've been trying to get google talk to work, but no luck yet: 1. when the jabber / google talk modules are loaded, asterisk ends up consuming all the CPU. This happens after a while (up to a day), not right after asterisk is (re-)started. 2. While i've been able to register a google talk account, I haven't been able to receive voice on it yet. I haven't had the time to file bug reports for these yet. regards, Stefan On 2/16/07, Kyle Sexton [EMAIL PROTECTED] wrote: Started playing with 1.4 and I'm curious what uses people have come up with for the Jabber integration? So far I can think of presence based call routing, but I'm sure there are other ideas. How are YOU using the new Jabber features in 1.4? :) -- Kyle Sexton ___ --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] Digium TE110P
Hello, Hmm, you are like me eons ago doing stuff on TE110P. Anyway, consider the following questions: 1. Is the jumper for that card set to E1 or T1? 2. Do you have an E1 or T1 link there? When you have an E1/T1 line there at your disposal, insert it in the slot. In either case, the green light should be steady and there will be no red light. Actually, the red light indicates that it ask for an active E1/T1 link. Hope it helps you. Regards, Demuel Hi I am currently installing a TE110P. SUSE10 The zttest test result is : average 99.9991%. My server : processor Intel® Celeron® D 330, 2.66 GHz, cache 256 Ko, FSB 533 MHz , 1G RAM. Hope it can help. Now I have a question to TE110P users : The card is physically plugged, modprobe, ztcfg ok etc ... There is a red led blinking. The question is : should this led provide a green continuous light or is it correct to have a red blinking light ? Thanks in advance Olivier Envoyez vos cartes de voeux depuis www.laposte.net Elles seront ensuite distribuées par le facteur : pratique et malin ! ___ --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] Summary of Trixbox vs. custom install
Hello everybody. First of all thanks to all the people giving their opinion on the subject I proposed: Trixbox vs. custom install. You've all been very helpful. I try to summarize what has emerged from the various messages. Forgive me if I miss or forget something or if I simplify too much some of your messages... - Elman Efendiyev says that you should install from sources if you need customized setup. He suggests using Slackware for Asterisk installations. - Lee Jenkins suggests using a GUI (like [EMAIL PROTECTED]), while admitting there are some problems with dialplan customization. He has also written his own Asterisk GUI to learn system internals, among other reasons. - Michael Collins suggests having two boxes two play with both Trixbox and a scratch install. Each method, he says, can teach a lot. He has still not decided which one to use in production. - Edward Halman suggests a step-by-step install. But he says also that if you just need Asterisk, FreePBX and A2Billing, Trixbox can be a good choice because setting up FreePPBX and A2Billing can be a little tricky. - Stephen Bosch writes: if you're only going to use Voip Trunks... use Trixbox. But if you're going to use PSTN hardware (like Digium or Sangoma cards), then a custom install is better. He had problems with Trixbox 2.0 and hardware, then replaced it with a custom install. He says also that the userbase of custom install is greater and has more advanced knowledge. - Shadowym states that if you're not able to set up Asterisk with a custom install, you should not use it in a production environment. - Mark Brooker says that both approaches are to be mastered. Anyway, you should be able to install from source to use Asterisk in a production environment. However, FreePBX is a great tool and should be used too. - Tom Rymes says that troubleshooting a GUI is much easier and Trixbox has no more problems with hardware than a custom install has. For example, on Sangoma site, there's a link to a customized Trixbox version targeted to Sangoma cards. Using Yum you can download new drivers an eventually install them. For what regarding the user base, Trixbox contains FreePBX and FreePBX has a huge user base, so that can also help. He concludes that if you want easy of use for you and your customers, you should use Trixbox. If you want complete control you should go for a custom install. - Tzafrir Cohen (in reply to Tom Rymes) reports problems with Yum update and says that abstraction can hide relevant details. For example, just to figure out if a FreePBX actually dialed, requires a trained Asterisk user examining the logs. - Stephen Bosch (in reply to Tom Rymes) says that he prefers not using binary distributions. About troubleshooting a GUI, he says that's not troubleshooting, it's more often debugging... Trixbox, furthermore, has little documentation. Furthermore, having to download drivers from various sites cancel the advantages of an easy Trixbox installation. And for what regarding the user base, he says that the messages regarding Trixbox are not answered so promptly within the Asterisk mailing list. He concludes: if you have at last to go back to pico/vim/emacs... better start with them. - Tzafrir Cohen (in reply to Stephen Bosch) suggests using SRPMS to rebuild packages from sources. - Tom Rymes (in reply to Tzafir Cohen) says he never had problems with Yum update. Of course you have to exclude the Kernel from the Yum updates. He reports installing Trixbox many times with Sangoma cards. He concludes that neither approaches, anyway, can be fine for everybody and you must choose the right approach according to your needs. I also include a consideration from mine: I would happily use Trixbox, because I did FreePBX setup once and it was a real pain, but I'm very frightened by a few issues: 1) Trixbox Macho installation that installs everything without asking. I, for example, would like to use software RAID (maybe it's wrong with Asterisk, but I want to do it!). I wouldn't like doing it manually after Trixbox installation. I would like to have an installer doing it for me. Centos (ex redhat) installer does it, so why Trixbox choose to install everything without prompting? 2) How easy it is to find Trixbox SRPMS? Is it possible to compile new software (i.e. Asterisk or FreePBX, etc..) on Trixbox without having to rewrite all the configuration files, changing all paths, all permissions, and so on... 3) Trixbox site and documentation are really SCARY. Someone should tell those guys. 4) How about updates? Are they published with a constant pace from the Trixbox team? Again (as for point 3) from the site and documentation I see, I don't expect me a very responsive development team. Thanks to everybody Stefano -- Stefano Corsi www.floo.it via della Fiera, 1 57029, Venturina - Campiglia Marittima (LI) Tel. 0565-836130 - Fax. 0565-836143 Cell.
[asterisk-users] Digium TE110P
Hi Demuel Thanks, it definitely helps a lot. I forgot to mention that I worked out the jumper thing. So, you give the explanation that is : as there is no E1/T1 connected to the card, the card is somewhat saying that it is waiting for a link. In this regard, it behaves differently from an analogic one, like the TDM400P, which shows a constant green light as soon as the driver is being loaded, even if there is no analogic link connected to the card. I was expecting the same behaviour from the TE110P. Thanks again, Olivier Envoyez vos cartes de voeux depuis www.laposte.net Elles seront ensuite distribuées par le facteur : pratique et malin ! ___ --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] freepbx with ASTERISK 1.4
it's possible to configure freepbx 2.2 with asterisk 1.4? Look here for the archives: http://lists.digium.com/pipermail/asterisk-users/ Search for the subject FreePBX 2.2.0 and Asterisk 1.4.0. You'll find EXACTLY what you're looking for. :-) Stefano ___ --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] Digium TE110P
Again, you need a E1/T1 link for that card and you need to set out the jumper either for T1 or E1 link. Since you are in Europe, the jumper settings should be for E1. This card is different from the TDM400P family. Regards, Demuel Hi Demuel Thanks, it definitely helps a lot. I forgot to mention that I worked out the jumper thing. So, you give the explanation that is : as there is no E1/T1 connected to the card, the card is somewhat saying that it is waiting for a link. In this regard, it behaves differently from an analogic one, like the TDM400P, which shows a constant green light as soon as the driver is being loaded, even if there is no analogic link connected to the card. I was expecting the same behaviour from the TE110P. Thanks again, Olivier Envoyez vos cartes de voeux depuis www.laposte.net Elles seront ensuite distribuées par le facteur : pratique et malin ! ___ --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] freepbx with ASTERISK 1.4
On Fri, 2007-02-16 at 09:33 -0500, McGhee, Stefano wrote: it's possible to configure freepbx 2.2 with asterisk 1.4? Look here for the archives: http://lists.digium.com/pipermail/asterisk-users/ Search for the subject FreePBX 2.2.0 and Asterisk 1.4.0. You'll find EXACTLY what you're looking for. :-) Look at: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.amportal.user/5377 Regards, Stefano ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] freepbx with ASTERISK 1.4
I said what to do before. http://freepbx.org/trac/ticket/1610 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillermo Salas M. Sent: 16 February 2007 14:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] freepbx with ASTERISK 1.4 On Fri, 2007-02-16 at 09:33 -0500, McGhee, Stefano wrote: it's possible to configure freepbx 2.2 with asterisk 1.4? Look here for the archives: http://lists.digium.com/pipermail/asterisk-users/ Search for the subject FreePBX 2.2.0 and Asterisk 1.4.0. You'll find EXACTLY what you're looking for. :-) Look at: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.amportal.user /5377 Regards, Stefano ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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
Re: [asterisk-users] Summary of Trixbox vs. custom install
Stefano Corsi wrote: Hello everybody. First of all thanks to all the people giving their opinion on the subject I proposed: Trixbox vs. custom install. You've all been very helpful. Very nice summary, Stefano. If you devote that kind of analysis to the question, you'll do fine, whatever you decide. -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] Digium TE110P
Hi, Thanks for info but could You please tell an exact mobel name of your motherboard? About led - when PRI cables not connected to TE100P or when there is a problem no physical level whth PRI led should be red blinking. When PRI link connecteg successfully led should provide a green continuous light -- Sincerely, Elman Efendiyev PROTECH INC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of rivoli.durand Sent: Friday, 16 February, 2007 15:52 To: asterisk-users Subject: [asterisk-users] Digium TE110P Hi I am currently installing a TE110P. SUSE10 The zttest test result is : average 99.9991%. My server : processor Intel® Celeron® D 330, 2.66 GHz, cache 256 Ko, FSB 533 MHz , 1G RAM. Hope it can help. Now I have a question to TE110P users : The card is physically plugged, modprobe, ztcfg ok etc ... There is a red led blinking. The question is : should this led provide a green continuous light or is it correct to have a red blinking light ? Thanks in advance Olivier Envoyez vos cartes de voeux depuis www.laposte.net Elles seront ensuite distribuées par le facteur : pratique et malin ! ___ --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] Re: Summary of Trixbox vs. custom install
As Stephen said, good summary. From my experience, installing from sources (with yum for updates and additional packages) I learned much about what is in the system. Frankly I did not find the GUIs to be ready for primetime when it comes to setting up a system. Using the GUI does not teach you about dialplans, SIP, or Zap configurations that are critical to understand if you are going to build/run a production system. Between google, this list, and some trial and error you should be able to get your first system up and running in a reasonable timeframe. The people here are experienced and very willing to help. Best Regards, Allen. ___ --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] 64 bit HPEC modules available?
On 2/16/07, Greg Siemon [EMAIL PROTECTED] wrote: I am running 64 bit linux on my Asterisk box and would like to get the new HPEC software running on it. However, while there are 32 bit modules available, there are no 64 bit modules on the ftp site: http://ftp.digium.com/pub/telephony/hpec/64-bit/ In some places on the digium website it states 32 bit only and other places including the documentation it states 32 64 bit are available. Is there a 64 bit version of the HPEC module or are the 32 bit modules suitable (can't imagine that they would be). Thanks in advance Greg ___ --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 I talked to tech support today... no 64bit yet. -- A.G. (Tony) Nichols I.S. Manager ___ --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 google talk
I also got the same problem on my Fedora Core 6, too. 2006/11/7, Mani Sridhar [EMAIL PROTECTED]: hi fellow asterisk enthusiasts, i've configured jabber.conf and gtalk.conf as descibed on voip-info.org (http://www.voip-info.org/wiki/view/Asterisk+Google+Talk). i see these messages on the CLI now, and i haven't been able to get Asterisk-Gtalk connectivity to work. *CLI [Nov 3 22:17:01] WARNING[30878]: res_jabber.c:1504 aji_recv_loop: JABBER: socket read error *CLI JABBER: gtalk_account OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='gmail.com' version='1.0' *CLI JABBER: gtalk_account INCOMING: ?xml version=1.0 encoding=UTF-8?stream:stream from=gmail.com id=D428120132AB91B7 version=1.0 xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client [Nov 3 22:17:01] ERROR[30878]: res_jabber.c:482 aji_act_hook: gnuTLS not installed. *CLI JABBER: gtalk_account INCOMING: stream:featuresstarttls xmlns=urn:ietf:params:xml:ns:xmpp-tls/mechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismX-GOOGLE-TOKEN/mechanism/mechanisms/stream:features *CLI these messages just keep appearing every 20s. gnuTLS is installed, so the error message gnuTLS not installed does not make sense to me. i checked config.log after running ./configure while building asterisk, and i can see that the check for gcc -lgnutls passed. [EMAIL PROTECTED] asterisk]# rpm -qi gnutls Name : gnutls Relocations: (not relocatable) Version : 1.0.25 Vendor: Red Hat, Inc. Release : 2.FC4 Build Date: Fri 10 Feb 2006 02:51:42 PM PST Install Date: Tue 31 Oct 2006 03:21:16 PM PST Build Host: hs20-bc1-7.build.redhat.com Group : System Environment/Libraries Source RPM: gnutls-1.0.25-2.FC4.src.rpm Size : 664600 License: LGPL Signature : DSA/SHA1, Fri 10 Feb 2006 05:10:47 PM PST, Key ID b44269d04f2a6fd2 Packager : Red Hat, Inc. http://bugzilla.redhat.com/bugzilla URL : http://www.gnutls.org/ Summary : A TLS implementation. Description : The GNU TLS library implements TLS. Someone needs to fix this description. [EMAIL PROTECTED] asterisk]# [EMAIL PROTECTED] asterisk]# ls -la /usr/lib/*gnutls* lrwxrwxrwx 1 root root 26 Oct 31 15:21 /usr/lib/libgnutls-extra.so.11 - libgnutls-extra.so.11.1.25 -rwxr-xr-x 1 root root 163832 Feb 10 2006 /usr/lib/libgnutls-extra.so.11.1.25 lrwxrwxrwx 1 root root 28 Oct 31 15:21 /usr/lib/libgnutls-openssl.so.11 - libgnutls-openssl.so.11.1.25 -rwxr-xr-x 1 root root 26756 Feb 10 2006 /usr/lib/libgnutls-openssl.so.11.1.25 lrwxrwxrwx 1 root root 20 Oct 31 15:22 /usr/lib/libgnutls.so - libgnutls.so.11.1.25 lrwxrwxrwx 1 root root 20 Oct 31 15:21 /usr/lib/libgnutls.so.11 - libgnutls.so.11.1.25 -rwxr-xr-x 1 root root 474012 Feb 10 2006 /usr/lib/libgnutls.so.11.1.25 [EMAIL PROTECTED] asterisk]# what can i check next? i'm pretty new (been working on asterisk for less than a month now) and i've been stuck at this point for a few days now. i'd really appreciate some pointers. thanks mani * Our reliance on access to a dialtone is now only slightly lesser than that on access to oxygen. _ Connect with your friends who use Yahoo! Messenger with Voice. Click! http://www.msnspecials.in/wlmyahoo/index.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 -- Best Regards Charles ___ --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
FW: [asterisk-users] Problem Transferring Direct to Voicemail
Could someone at least respond to this so that I know it is getting out
there? I have posted this three times and not gotten one single
response. I even totally reworded it hoping that would help.
I'm at a loss here and not sure where to turn next. All searches I've
done come up with nothing telling me what Notify answer on an owned
channel means and what to do about it.
PLEASE!! Someone?? Anyone???
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Savoy,
Kevin - Williston, ND
Sent: Wednesday, February 14, 2007 8:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Problem Transferring Direct to Voicemail
I am having an issue with 1.4 where we can't successfully transfer a
call directly to a voicemail box. We hit Transfer on the phone and
dial the mailbox number we want to send it to,
My dial plan for this is:
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
The voicemail system picks up and starts to play its message and at this
point. We should then hit Transfer again at this point the person
doing the transfer should drop off the call. However we just continue to
hear the voicemail message and the caller continues to sit on hold.
On the Asterisk CLI I see the following:
[Feb 9 11:52:03] WARNING[5054]: chan_sip.c:12310 handle_response:
Notify answer on an owned channel?
Can anyone tell me what this means or how to fix it?
Please help.
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 http://www.novo1.com/
Novo 1 is a service mark of Novo 1, Inc
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Re: [asterisk-users] Summary of Trixbox vs. custom install
On Feb 15, 2007, at 7:01 PM, Stefano Corsi wrote: Hello everybody. First of all thanks to all the people giving their opinion on the subject I proposed: Trixbox vs. custom install. You've all been very helpful. [snip] I also include a consideration from mine: I would happily use Trixbox, because I did FreePBX setup once and it was a real pain, but I'm very frightened by a few issues: 1) Trixbox Macho installation that installs everything without asking. I, for example, would like to use software RAID (maybe it's wrong with Asterisk, but I want to do it!). I wouldn't like doing it manually after Trixbox installation. I would like to have an installer doing it for me. Centos (ex redhat) installer does it, so why Trixbox choose to install everything without prompting? Stefano, Great summary. As an aside here, it is possible to install Trixbox on top of an existing CentOS installation by using the tarball, not the ISO. This works very well, with one issue I ran into. A fresh install of CentOS updated via yum will not have the correct version of the kernel to match the zaptel-modules RPM shipped with Trixbox (because it is no longer in the repositories). You can fix this problem two ways: 1.) Manually install the kernel from the Trixbox CD, which will fix the problem, if you prefer to work just the way Trixbox normally does. You should configure yum to not upgrade the kernel in this case, because that would break zaptel. 2.) You can download and manually recompile zaptel on your own. Either you will have to recompile zaptel every time that the kernel is upgraded by yum, or you should configure yum to not upgrade the kernel. (This is true of any zaptel install, not just Trixbox.) See the bug i posted: http://www.trixbox.org/modules/xproject/ index.php?op=viewTicketMainid=27 Another resolution would be to provide an SRPM for the zaptel-modules package, which you (or the tarball install script) could rpmbuild -- rebuild against your current kernel. Either way, this isn't a big problem so long as you know it's there. Worst case scenario, you just download and compile zaptel, which you would have had to do anyway for a non-trixbox install. Tom ___ --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] Debugging a SIP / AudioCodes Problem
Out of curiosity, want to know what GPL violations did AudioCodes do and in which products ? Thanks, Prasad. Andrew D Kirch [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: Audiocodes blatently violates the GPL... dont use their gear. On 11 Feb 2007 19:11:51 -, [EMAIL PROTECTED] wrote: I have 2 identical AudioCodes MP-112s. They have the same config except for the SIP usernames/passwords and the device IP. The configs in extension.conf and sip.conf are also identical. On one box, when I pick up the phone, I get a fast busy and the logs/debug show an automatic hangup. On the other device, I can make calls without a problem. I can even call the phone that can't make a call. Any ideas where I could start to figure out where the problem is? ___ --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 Indeed step 1: throw the audiocodes in the trash step 2: buy real hardware -- Andrew D Kirch | Abusive Hosts Blocking List | www.ahbl.org Security Admin | Summit Open Source Development Group | www.sosdg.org Key fingerprint = 4106 3338 1F17 1E6F 8FB2 8DFA 1331 7E25 C406 C8D2 ___ --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 - The fish are biting. Get more visitors on your site using Yahoo! Search Marketing.___ --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: FW: [asterisk-users] Problem Transferring Direct to Voicemail
I am having an issue with 1.4 where we can't successfully transfer a call
directly to a voicemail box. We hit Transfer on the phone and dial the
mailbox number we want to send it to,
My dial plan for this is:
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
The voicemail system picks up and starts to play its message and at this
point. We should then hit Transfer again at this point the person doing
the transfer should drop off the call. However we just continue to hear the
voicemail message and the caller continues to sit on hold.
I've not worked with 1.4 much yet, but I'd try changing my dialplan to:
exten=_*40XX,1,Answer
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
That way, I would know that the channel is answered, which is what often
will stop IP phones from allowing the attended transfer to complete.
Hope that helps,
David
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[asterisk-users] MixMonitor RingBack Tone Issue
Hi,
I use in Production : Asterisk 1.2.9.1
We Use Asterisk as a SIP Transit Server to record centrally all the calls.
The call flow would be:
incoming calls : PSTN - GW -SIP- Asterisk(Record) -SIP- Softswitch - IP
Phone
outgoing calls : IP Phone - Softswitch -SIP- Asterisk(Record) -SIP- GW
- PSTN
Dial plan in Asterisk is quite simple:
[record]
exten = s,1,Set(CALLFILENAME=${TIMESTAMP}-${UNIQUEID})
exten = s,n,Set(CALLERID(name)=${CALLERID(name)})
exten = s,n,Set(CALLERID(number)=00${CALLERID(number)})
exten = s,n,MixMonitor(${CALLFILENAME}.WAV,b)
exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r)
Everything works perfectly, except when the softswitch, or the PSTN sends
back RingBack Tone.
I can see the RTP flow arriving to Asterisk,
but, it seems that Asterisk doesn't forward it to the other party
(next-hop).
Any ideas why ?
How can I bypass this issue ?
Thanks,
Jean-Marc
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Re: FW: [asterisk-users] Problem Transferring Direct to Voicemail
Savoy, Kevin - Williston, ND wrote: Could someone at least respond to this so that I know it is getting out there? I have posted this three times and not gotten one single response. I even totally reworded it hoping that would help. I’m at a loss here and not sure where to turn next. All searches I’ve done come up with nothing telling me what Notify answer on an owned channel means and what to do about it. PLEASE!! Someone?? Anyone??? Bonjour. You might consider going back to a 1.2.x version if this is for a production system. -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
[asterisk-users] Distinct call permissions for each user
Dear all, How may I configure my extensions.conf to stablish different PSTN access permissions for each user, letting for example user_A make only local calls and user_B make local and long-distance calls? I guess it can be done using include of other contexts, but how exactly? someone please give me one example? Thanks in advance, Regards, Ricardo. ___ --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 callerID
From: voip crazy [EMAIL PROTECTED] Date: Fri, 16 Feb 2007 13:28:26 +0100 Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I make a call directly using one of this line, the callerID is sending correctly. With the same zapata config file and the Freepbx 2.1.3, the callerId was sending correctly. This is a little confusing. To which phones callerID is not delivered? Phones out on the PSTN/mobile via FXO? I don't believe that Asterisk (or any device for that matter) can deliver callerID via an FXO, which by definition is an alalogue interface. A callerID is associated with the FXS in the central office that connects to your FXO at the switch side. On the other hand, Asterisk should deliver callerID to most (if not all) digital circuits, be them VoIP or PSTN. Yuan Liu Any clue will be welcome Thanks in advance. VoipCrazy -- zapata.conf-- [channels] language=en context=from-zaptel rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes callerid=asreceived hidecallerid=no callwaiting=yes usecallingpres=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=50 immediate=no rxgain=3.0 txgain=4.0 immediate=no busydetect=yes busycount=8 answeronpolarityswitch=yes hanguponpolarityswitch=yes ringtimeout=8000 faxdetect=both signalling=fxs_ks useincomingcalleridonzaptransfer=yes channel = 1-2 ___ --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] sangoma 102 and CAB-E1-RJ45BNC
Hi, sorry for the newbie hardware questions but here it goes scenario - our telco is feeding us e1 thru coax connection (unbalanced) - so the coax feed rx-tx goes to our old pabx using ericsson bp250 - what we wanted to do is to install asterisk in between hence telco--asterisk--bp250 using asterisk to power up the voip portion the problem is the we are getting crackling sound when we make calls from the old pabx extension, it seems that there is a lot of line noise due to emc. so here goes the newbie question: current setup is that from the coax we are using a balun using the given cables from sangoma will the cisco *CAB-E1-RJ45BNC *connector work on the 102 ie no need to use the balun ___ --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] Sangoma A101 install problem
I just got a brand new A101 and am trying to install it in my test Asterisk box. The install went without a hitch. I followed the directions on the Sangoma Wiki: Wanpipe Asterisk Install http://sangoma.editme.com/wanpipe-linux-asterisk-install Wanpipe for Asterisk Configuration http://sangoma.editme.com/wanpipe-asterisk-configure It went perfect, no problems and Asterisk came up fine. I downed the box and moved it from my office to the computer room where I could hook it up to a test T1. I booted the box up and it looked like Zaptel wasn't installed. I went to the zaptel source directory and did a make clean, make install no errors, but now ztcfg -vvv shows this: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: E M (Default) (Slaves: 01) Channel 02: E M (Default) (Slaves: 02) Channel 03: E M (Default) (Slaves: 03) Channel 04: E M (Default) (Slaves: 04) Channel 05: E M (Default) (Slaves: 05) Channel 06: E M (Default) (Slaves: 06) Channel 07: E M (Default) (Slaves: 07) Channel 08: E M (Default) (Slaves: 08) Channel 09: E M (Default) (Slaves: 09) Channel 10: E M (Default) (Slaves: 10) Channel 11: E M (Default) (Slaves: 11) Channel 12: E M (Default) (Slaves: 12) Channel 13: E M (Default) (Slaves: 13) Channel 14: E M (Default) (Slaves: 14) Channel 15: E M (Default) (Slaves: 15) Channel 16: E M (Default) (Slaves: 16) Channel 17: E M (Default) (Slaves: 17) Channel 18: E M (Default) (Slaves: 18) Channel 19: E M (Default) (Slaves: 19) Channel 20: E M (Default) (Slaves: 20) Channel 21: E M (Default) (Slaves: 21) Channel 22: E M (Default) (Slaves: 22) Channel 23: E M (Default) (Slaves: 23) Channel 24: E M (Default) (Slaves: 24) Channel 25: FXS Kewlstart (Default) (Slaves: 25) 25 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) Any ideas what went wrong? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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
[asterisk-users] Experiences with FoneBridge2 / TDMoE?
I'm scoping out HA for a relatively simple Office/Call Center PBX. Current setup uses a TE412P with 4 PRI our telco with SIP hard/soft phones for users. Some outbound also goes to a SIP provider. Active/Active looks to be too much hassle for an installation this size, so we're looking at adding an extra * in an active/passive configuration with Linux-HA in between them. Does anyone have any experience using The Redfone Fonebridge2 in this configuration? It seems like it would do the trick, but I can only find the one testimonial/tutorial on voip-info.org. Specifically, I'm wondering about the reliability of the device itself (since you can't seem to pair them in any way) as well as what extra work the * box has to do in a TDMoE configuration. I can't find any mention of EC being done on the Fonebridge, so I assume you'd have to use software echo cancellation in the zaptel driver. Other than the CPU to do that, does zaptel take up more/less CPU reading frames from a PCI card vs TDMoE? Any experience or suggestions on other ways to do this are appreciated. -- j. ___ --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] Summary of Trixbox vs. custom install
Actually, there's a very easy way to install Trixbox with RAID right from the CD. All you have to do is edit one file on the root of the ISO, burn the image and boot from it. I have used it myself with great success, though I'm not sure if it has been tested on 2.0. The instructions are at http://dumbme.voipeye.com.au/trixbox/trixbox_without_tears.htm#_Toc15759 1311 John McCollough LAN Network Connections, Inc (603)622-8557 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Friday, February 16, 2007 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Summary of Trixbox vs. custom install On Feb 15, 2007, at 7:01 PM, Stefano Corsi wrote: Hello everybody. First of all thanks to all the people giving their opinion on the subject I proposed: Trixbox vs. custom install. You've all been very helpful. [snip] I also include a consideration from mine: I would happily use Trixbox, because I did FreePBX setup once and it was a real pain, but I'm very frightened by a few issues: 1) Trixbox Macho installation that installs everything without asking. I, for example, would like to use software RAID (maybe it's wrong with Asterisk, but I want to do it!). I wouldn't like doing it manually after Trixbox installation. I would like to have an installer doing it for me. Centos (ex redhat) installer does it, so why Trixbox choose to install everything without prompting? Stefano, Great summary. As an aside here, it is possible to install Trixbox on top of an existing CentOS installation by using the tarball, not the ISO. This works very well, with one issue I ran into. A fresh install of CentOS updated via yum will not have the correct version of the kernel to match the zaptel-modules RPM shipped with Trixbox (because it is no longer in the repositories). You can fix this problem two ways: 1.) Manually install the kernel from the Trixbox CD, which will fix the problem, if you prefer to work just the way Trixbox normally does. You should configure yum to not upgrade the kernel in this case, because that would break zaptel. 2.) You can download and manually recompile zaptel on your own. Either you will have to recompile zaptel every time that the kernel is upgraded by yum, or you should configure yum to not upgrade the kernel. (This is true of any zaptel install, not just Trixbox.) See the bug i posted: http://www.trixbox.org/modules/xproject/ index.php?op=viewTicketMainid=27 Another resolution would be to provide an SRPM for the zaptel-modules package, which you (or the tarball install script) could rpmbuild -- rebuild against your current kernel. Either way, this isn't a big problem so long as you know it's there. Worst case scenario, you just download and compile zaptel, which you would have had to do anyway for a non-trixbox install. Tom ___ --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] Jabber/Asterisk Integration
Kyle Sexton wrote: Started playing with 1.4 and I'm curious what uses people have come up with for the Jabber integration? So far I can think of presence based call routing, but I'm sure there are other ideas. How are YOU using the new Jabber features in 1.4? :) We've been using it since July last year (brave / stupid - make your choice) for integrating our custom application with the asterisk system. The phone system sends all sorts of call information to the agent about to receive the call, whilst the agent monitoring screen is used to monitor the presence of the agents and their dialplan status (dialling / calling / etc etc) Julian. ___ --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 can I use 'Asterisk Manager API' to hold and retrive an active call?
Thanks Stefan for input.
I know that there is a hangup action in Asterisk Manager API.
I am looking for hold and retrive commend. I search google and find
that redirecting to parkslot can work.
If I have a PSTN call connecting to Asterisk and then to a SIP
extension, there are two connections here. If I redirect one channel to
parkslot, another channel will automatically hangup.
Later, if I redirect that channel from parkslot to the SIP extension,
that extension will ring again.
Is there a solution to redirect that two channels to parkslot at the
same time, then reconnect these two channels without ringing?
-Original Message-
James Zhang wrote:
These are common functions. Why Asterisk Manager
doesn't provide commands to hold and retrive an active channel?
If it must be implemented by AGI, could anyone give a direction or
steps?
Sure the Manager API provides all thing to do that.
Maybe you are just using the wrong library on top of the Manager API ;)
Asterisk-Java as an example lets you retrieve active channels, iterate
over them, hangup, redirect, ... whatever.
Example to hangup all active channels:
for (AsteriskChannel channel : server.getChannels())
{
channel.hangup();
}
http://asterisk-java.org
I am sure other libraries provide similar abstraction.
=Stefan
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[asterisk-users] Re: Summary of Trixbox vs. custom install
To piggy-back off of what Allen said, much of what I have learned about configuring Asterisk and working with Linux has come from constructing my system the manual way. I use FC5, but I avoid using yum and don't install from rpms when I can avoid it. I typically install everything I need from sources because I can load the modules I need and I know (or can specify) which directories the binaries, scripts and config files go to. And in troubleshooting, I know where to look. I seem to only have problems when I take the rpm shortcut. Thanks to this list, I can get around dialplans and the SIP config files with confidence. I make extensive use of AGI (php and MySQL) in my business application, all thanks to people on this list. FreePBX was a great beginning, but for me, that's all it was good for, a beginning. I went through a similar metamorphosis with learning to configure postfix and dovecot. There is a similar freepbx-like web gui for configuring a mail server that I used in the beginning as well. I am a total Linux/Asterisk newbie and the process has been full of growing pains, but I am glad I went through it. I owe this list a lot, and of course a very patient employer who went through many system crashes with me without pulling his hair out or complaining because the phones were down. Ed Halman (718) 705-7451 [EMAIL PROTECTED] -- Message: 26 Date: Fri, 16 Feb 2007 11:05:12 -0500 From: Allen Casteran [EMAIL PROTECTED] Subject: [asterisk-users] Re: Summary of Trixbox vs. custom install To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed As Stephen said, good summary. From my experience, installing from sources (with yum for updates and additional packages) I learned much about what is in the system. Frankly I did not find the GUIs to be ready for primetime when it comes to setting up a system. Using the GUI does not teach you about dialplans, SIP, or Zap configurations that are critical to understand if you are going to build/run a production system. Between google, this list, and some trial and error you should be able to get your first system up and running in a reasonable timeframe. The people here are experienced and very willing to help. Best Regards, Allen. -- ___ --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] Pickup application
did you use correct context to pickup external call?
if you simply pickup without context, it will try to pickup ringing line
in @from-internal context, from you example...
PJ
nik600 wrote:
I am trying to configure the pickup.
This is my dialplan:
exten = _57.,1,Pickup(${EXTEN:2})
So, when i call for example 57333 Asterisk tries to pick up the call
ringing on 333
The problem is that it works only with internal calls!
For example, if i call 333 from 334 and while 333 i ringing i try to
dial 57333 it works.
If i call an external number that via dialplan dials 333 dialing 57333
i got:
Executing [EMAIL PROTECTED]:1] Pickup(SIP/200-08432290, 333) in
new stack
[Feb 16 09:20:46] NOTICE[7586]: app_directed_pickup.c:159 pickup_exec:
No target channel found for 333.
Where am i wrong?
Thanks
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RE: [asterisk-users] Sangoma A101 install problem
I reran the install and I had answered one question wrong. I think this fixed it. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, February 16, 2007 2:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Sangoma A101 install problem I just got a brand new A101 and am trying to install it in my test Asterisk box. The install went without a hitch. I followed the directions on the Sangoma Wiki: Wanpipe Asterisk Install http://sangoma.editme.com/wanpipe-linux-asterisk-install Wanpipe for Asterisk Configuration http://sangoma.editme.com/wanpipe-asterisk-configure It went perfect, no problems and Asterisk came up fine. I downed the box and moved it from my office to the computer room where I could hook it up to a test T1. I booted the box up and it looked like Zaptel wasn't installed. I went to the zaptel source directory and did a make clean, make install no errors, but now ztcfg -vvv shows this: Zaptel Version: 1.4.0 Echo Canceller: MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: E M (Default) (Slaves: 01) Channel 02: E M (Default) (Slaves: 02) Channel 03: E M (Default) (Slaves: 03) Channel 04: E M (Default) (Slaves: 04) Channel 05: E M (Default) (Slaves: 05) Channel 06: E M (Default) (Slaves: 06) Channel 07: E M (Default) (Slaves: 07) Channel 08: E M (Default) (Slaves: 08) Channel 09: E M (Default) (Slaves: 09) Channel 10: E M (Default) (Slaves: 10) Channel 11: E M (Default) (Slaves: 11) Channel 12: E M (Default) (Slaves: 12) Channel 13: E M (Default) (Slaves: 13) Channel 14: E M (Default) (Slaves: 14) Channel 15: E M (Default) (Slaves: 15) Channel 16: E M (Default) (Slaves: 16) Channel 17: E M (Default) (Slaves: 17) Channel 18: E M (Default) (Slaves: 18) Channel 19: E M (Default) (Slaves: 19) Channel 20: E M (Default) (Slaves: 20) Channel 21: E M (Default) (Slaves: 21) Channel 22: E M (Default) (Slaves: 22) Channel 23: E M (Default) (Slaves: 23) Channel 24: E M (Default) (Slaves: 24) Channel 25: FXS Kewlstart (Default) (Slaves: 25) 25 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) Any ideas what went wrong? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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
[asterisk-users] Re: Asterisk callerID
voip crazy wrote: Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I make a call directly using one of this line, the callerID is sending correctly. With the same zapata config file and the Freepbx 2.1.3, the callerId was sending correctly. Any clue will be welcome Again, your description is not clear. If your problem is that calls coming in to asterisk are not displaying caller ID on your phones, then you need to make sure that your CO lines are configured by the carrier to deliver caller ID. As a test connect a basic analog phone that has caller ID capability and call the line. If your simple phone displays the CallerID that you are calling from your line supports it and Asterisk should pick it up. If you do not see the caller ID on the analog phone when directly connected to the CO line, then call your carrier and ask them to provide Caller ID on your lines. I had this exact situation this morning, so yes it happens. If your problem is calling OUT from asterisk and your caller ID not getting displayed on the phone you are calling, that is also a function of the carrier and something you have NO control over. You should see something on the far end even if its Private unavailable or blocked. Call your carriers and ask them to check their set up for your phone lines. Allen. ___ --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] Open Source VoIP at FOSDEM
For those of you coming to FOSDEM on 24/25 Feb, there'll be a session in the Debian devroom on Open Source VoIP. http://www.fosdem.org/2007/schedule/speakers/daniel+pocock Several VoIP projects will be represented in various ways throughout the weekend, and there will be some of the following: - hardware giveaways from leading VoIP companies - launch of new open source VoIP product during the session in the Debian devroom - integration of VoIP features into other applications (e.g. OpenGroupware) will also be discussed and demonstrated I look forward to seeing some of you there. Regards, Daniel ___ --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] Re: PSTN Calls from SI.P: buzzing and pops
Allen Casteran wrote: Tzafrir Cohen wrote: Any change if you run: echo 0 0 /proc/xpp/sync Tzafrir, Yes, that was it. Problem solved. Thanks again for your help. Allen. ___ --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: FW: [asterisk-users] Problem Transferring Direct to Voicemail
Maybe nobody knows. I certainty know that I've never ever seen that error. Savoy, Kevin - Williston, ND wrote: Could someone at least respond to this so that I know it is getting out there? I have posted this three times and not gotten one single response. I even totally reworded it hoping that would help. I'm at a loss here and not sure where to turn next. All searches I've done come up with nothing telling me what Notify answer on an owned channel means and what to do about it. PLEASE!! Someone?? Anyone??? ___ --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] iaxmodem - fax tone?
I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax server and having it then dial using iaxmodem is working fine. Hylafax server is talking to my Asterisk box that has a Sangoma A101 in it via iaxmodem via an IAX channel using ulaw. A call coming into a certain test DID comes into the Sangoma A101 then it goes to another box via IAX ulaw that uses the rxfax app to capture the fax. I am using rxfax right now because I am just starting to test hylafax. Normal faxes into my test DID work fine so I know it's not the Sangoma or rxfax app or communication between those 2 servers. I notice the Sangoma detects incoming fax tones for these faxes. However, if I send the fax via iaxmodem, I notice the Sangoma sends the call out via my trunk group that turns off the echo cancellation (Because I am using a group I setup to do that, there it's surprise there) but when it comes back in, the Sangoma is not turning off echo cancellation. The end result is rxfax gets the tiff image, but then the call just hangs up and the pdf is never created and sent. The Tiff image appears to be complete however but the call just hangs up after that. I think the problem is due to the Sangoma not detecting the fax tones. Am I missing a setting with iaxmodem or hylafax? To recap: Test DID works fine with normal analog and other fax via ATA adapters so I think I can safely rule out a misconfiguration there Iaxmodem registered, hylafax clearly sends the fax via it as I see it coming back in and the tiff created using rxfax The problem appears to be coming back - echo cancellation not being turned off. Unfortunately like many of us, I don't have a test PRI server I can play with so I have to do this after hours. I will be turning off echo cancellation late at night and seeing if that solves the problem but wanted to pose this question to the list. I have _not_ tested it to an outside analog fax yet via hylafax/iaxmodem. Bill ___ --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: FW: [asterisk-users] Problem Transferring Direct to Voicemail
Well thanks to those who did reply. I guess I'll have to live with it until somehow it gets fixed. The reason I upgraded to 1.4 is that there were three or four other issues I had that this fixed. Going back just isn't really an option since those issues were bigger then this one. Guess we'll live with it for now. If anyone ever hears of this and a fix for it please let me know. Again thanks for responding this time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Friday, February 16, 2007 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: FW: [asterisk-users] Problem Transferring Direct to Voicemail Maybe nobody knows. I certainty know that I've never ever seen that error. Savoy, Kevin - Williston, ND wrote: Could someone at least respond to this so that I know it is getting out there? I have posted this three times and not gotten one single response. I even totally reworded it hoping that would help. I'm at a loss here and not sure where to turn next. All searches I've done come up with nothing telling me what Notify answer on an owned channel means and what to do about it. PLEASE!! Someone?? Anyone??? ___ --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] Open Source VoIP at FOSDEM
This reminds me, we are still looking for some one or some company to step up and take charge of the VOIP track of sessions at BarCampUSA in August. There have been a number of people showing interest in speaking and exhibiting at the event but so far no one has come forward to chair the whole series of discussions. Due to the large percentage of the 5000+ expected people I am imagining that VOIP will be one of the largest components. More information available at www.BarCampUSA.org or give me a call on the numbers below or via Corraleta Connect at http://www.cognation.net/contact Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Daniel Pocock Sent: Friday, 16 February 2007 3:21 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Open Source VoIP at FOSDEM For those of you coming to FOSDEM on 24/25 Feb, there'll be a session in the Debian devroom on Open Source VoIP. http://www.fosdem.org/2007/schedule/speakers/daniel+pocock Several VoIP projects will be represented in various ways throughout the weekend, and there will be some of the following: - hardware giveaways from leading VoIP companies - launch of new open source VoIP product during the session in the Debian devroom - integration of VoIP features into other applications (e.g. OpenGroupware) will also be discussed and demonstrated I look forward to seeing some of you there. Regards, Daniel ___ --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 Transferring Direct to Voicemail
Two Things,
#1, You do not say what model of telephone you are using. You should also
mention software version and if you can transfer calls to other locations.
#2, Have you tried a SIP debug?
I don't see why this would matter but
I don't see your entire dialplan and I don't see a priority #1.
Do you have other applications at this pattern match? Otherwise try
replacing the ,n, with ,1,
What does the following command show you in the CLI:
' show dialplan [EMAIL PROTECTED] ' -- replace the context with whatever
you are using.
Eric Osterberg
Sound Choice Communications LLC
- Minnesota, US
On Wed, 14 Feb 2007, Savoy, Kevin - Williston, ND wrote:
I am having an issue with 1.4 where we can't successfully transfer a
call directly to a voicemail box. We hit Transfer on the phone and
dial the mailbox number we want to send it to,
My dial plan for this is:
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
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[asterisk-users] DNIS on T1 channels
I installed a Sangoma card with the default install. I'm getting five digits of DNIS with each call. The T1 is setup ESF/B8ZS wink start. Each of the digits of the DNIS are being used for extensions in the context. I need a single extension that let me start an AGI script that can use the dnis. Can anyone point me in the right direction to do this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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
[asterisk-users] MINNESOTA: TwinCities Asterisk Users Group - Saturday February 17th 2007 - 11:00am
This is a reminder that the Twin Cities Asterisk Users Group will be meeting this Saturday, Feb 17 at 11:00am. - This month's meeting is primarily a business meeting to discuss the agenda for the coming year. Last weekend I was unable to present or host the meeting in my offices because I had another appointment. We need other locations and presenters. If you have a topic you would like to see presented, this is your chance to have your say for this coming year. Meetings are normally held monthly on the second Saturday of each month, excluding July and December. The Agenda is posted online http://www.voip-info.org/wiki/index.php?page=Twin+Cities+Asterisk+User+Group+Agenda This meeting will be held at Sound Choice Communications LLC in Bloomington, MN... Hope to see you at: 11:00 AM - Saturday Sound Choice Communications LLC 7839 12th Ave So Bloomington MN 55425 +1.(651)-999-0888 -Eric Osterberg ___ --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] End Wrap-up Time?
I patched 1.4.0 to add a command to the manager api in the queue application to implement the end wrap-up time I was asking about. All the command does is modify the 'lastcall' timestamp for the queue member by subtracting the value of the queue's defined wrapup time. Andrew Kohlsmith wrote: On 2/15/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: This is coming right out of left field, as I've never set up an Asterisk queue or agent system, but is it possible to pause and unpause while in the wrap-up time? What happens? Does the wrapup time go away then? Might be a counter-intuitive way around it if so... On Thursday 15 February 2007 4:34 pm, Matt wrote: I tried that. It didn't work :( What if we patched Asterisk to do just that? What could the repercussions be? They're already pausing/unpausing, so having the wrapup time auto-zero on unpause seems a non-issue... -A. ___ --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 Transferring Direct to Voicemail
The phones are Polycom 501's. I did confirm that this does work with
1.2.9.1 and not in 1.4. I upgraded to 1.4 because it fixed other issues
such as transferring calls out to an external number and echo issues.
I didn't have the entire dial plan because I didn't think it would
matter either. I do have a priority 1 to do an Answer() first and then
the n priority.
Here is the dial plan.
exten=_*40XX,1,Answer()
exten=_*40XX,n,Wait(2)
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
exten=_*40XX,n,Hangup()
Below is what I get at the CLI
wpbx1*CLI dialplan show [EMAIL PROTECTED]
[ Included context 'inbound' created by 'pbx_config' ]
'_*40XX' = 1. Answer()
[pbx_config]
2. Wait(2)
[pbx_config]
3. Voicemail(${EXTEN:1}|u)
[pbx_config]
4. Hangup()
[pbx_config]
I also get the same thing when I do a dialplan show *4033@ with the
context inbound, outbound and default. I have tried moving the
dialplan portion above to many different contexts, from the default to
the office context where the phones are to the inbound and outbound
contexts but I always get that owned channel? message that I referred
too.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, February 16, 2007 3:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem Transferring Direct to Voicemail
Two Things,
#1, You do not say what model of telephone you are using. You should
also
mention software version and if you can transfer calls to other
locations.
#2, Have you tried a SIP debug?
I don't see why this would matter but
I don't see your entire dialplan and I don't see a priority #1.
Do you have other applications at this pattern match? Otherwise try
replacing the ,n, with ,1,
What does the following command show you in the CLI:
' show dialplan [EMAIL PROTECTED] ' -- replace the context with whatever
you are using.
Eric Osterberg
Sound Choice Communications LLC
- Minnesota, US
On Wed, 14 Feb 2007, Savoy, Kevin - Williston, ND wrote:
I am having an issue with 1.4 where we can't successfully transfer a
call directly to a voicemail box. We hit Transfer on the phone and
dial the mailbox number we want to send it to,
My dial plan for this is:
exten=_*40XX,n,Voicemail(${EXTEN:1},u)
___
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[asterisk-users] Help needed to server code on Vxworks
Folks, How much efforts are needed to make Asterisk code to run on Vxworks? Is there any document in the distribution which describes the steps to follow to run on Vxworks. Is there any limitation in Vxworks which should be disabled or remove in Asterisk server code. Thanks, Murali ___ --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] Does Asterisk support DNIS?
The subject pretty much says it all. Does Asterisk support DNIS, and if so, what kind of connection is required? (T1, PRI) I've got a wink start T1. I've read comments that say the DNIS will be seen as an extension, but I'm seeing each digit of the DNIS as a separate extension. So in my case I send DNIS of 12345, Asterisk will jump from extension 1 to extension 2 to extension 3 to extension 4 to extension 5. Only executing the first one or two lines in each. This is a PITA! And make absolutely no sense to me. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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
Re: [asterisk-users] End Wrap-up Time?
Where is this patch? On 2/16/07, James Fromm [EMAIL PROTECTED] wrote: I patched 1.4.0 to add a command to the manager api in the queue application to implement the end wrap-up time I was asking about. All the command does is modify the 'lastcall' timestamp for the queue member by subtracting the value of the queue's defined wrapup time. Andrew Kohlsmith wrote: On 2/15/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: This is coming right out of left field, as I've never set up an Asterisk queue or agent system, but is it possible to pause and unpause while in the wrap-up time? What happens? Does the wrapup time go away then? Might be a counter-intuitive way around it if so... On Thursday 15 February 2007 4:34 pm, Matt wrote: I tried that. It didn't work :( What if we patched Asterisk to do just that? What could the repercussions be? They're already pausing/unpausing, so having the wrapup time auto-zero on unpause seems a non-issue... -A. ___ --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] DNIS on T1 channels
I don't totally understand your question. * Your T1 is providing DNIS. * You are receiving the DNIS * Add a line to your from-pstn, from-trunk, or whatever from your T1 is called that when it sees those 5 digits you want it runs the AGI. On 2/16/07, David Ruggles [EMAIL PROTECTED] wrote: I installed a Sangoma card with the default install. I'm getting five digits of DNIS with each call. The T1 is setup ESF/B8ZS wink start. Each of the digits of the DNIS are being used for extensions in the context. I need a single extension that let me start an AGI script that can use the dnis. Can anyone point me in the right direction to do this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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] iaxmodem - fax tone?
From: Bill Gibbs [EMAIL PROTECTED] Date: Fri, 16 Feb 2007 15:55:13 -0500 I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax server and having it then dial using iaxmodem is working fine. Hylafax server is talking to my Asterisk box that has a Sangoma A101 in it via iaxmodem via an IAX channel using ulaw. A call coming into a certain test DID comes into the Sangoma A101 then it goes to another box via IAX ulaw that uses the rxfax app to capture the fax. I am using rxfax right now because I am just starting to test hylafax. Normal faxes into my test DID work fine so I know it's not the Sangoma or rxfax app or communication between those 2 servers. I notice the Sangoma detects incoming fax tones for these faxes. However, if I send the fax via iaxmodem, I notice the Sangoma sends the call out via my trunk group that turns off the echo cancellation (Because I am using a group I setup to do that, there it's surprise there) but when it comes back in, the Sangoma is not turning off echo cancellation. The end result is rxfax gets the tiff image, but then the call just hangs up and the pdf is never created and sent. The Tiff image appears to be complete however but the call just hangs up after that. I don't have experience with iaxmodem (although your posting just gave me great confidence that I could use it). But you missed one piece of info: when normal FAX' send to your test DID, does Sangoma turn off echo cancellation on that channel? This would be a useful test to confirm your theory that echo cancellation is causing the problem. Another piece of useful information would be, when you say when it comes back in, do you mean you are using iaxmodem to dial another channel in the trunk group it dials out, and that channel uses iaxmodem on the same server to receive the FAX call? Yet another important - and easy - test would be the one you haven't done: send a FAX via the trunk to an alalogue FAX. Or may be you have? I mean, send a FAX via Sangoma to an alalogue line to receive the FAX. If you are concerned about spamming other people's FAX machine, you can even set up eFAX for free in one minute, and send the test FAX to yourself. (But get to wonder how your office FAX is connected.:-) Yuan Liu I think the problem is due to the Sangoma not detecting the fax tones. Am I missing a setting with iaxmodem or hylafax? To recap: Test DID works fine with normal analog and other fax via ATA adapters so I think I can safely rule out a misconfiguration there Iaxmodem registered, hylafax clearly sends the fax via it as I see it coming back in and the tiff created using rxfax The problem appears to be coming back - echo cancellation not being turned off. Unfortunately like many of us, I don't have a test PRI server I can play with so I have to do this after hours. I will be turning off echo cancellation late at night and seeing if that solves the problem but wanted to pose this question to the list. I have _not_ tested it to an outside analog fax yet via hylafax/iaxmodem. Bill ___ --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] Does Asterisk support DNIS?
From: David Ruggles [EMAIL PROTECTED] Date: Fri, 16 Feb 2007 17:02:38 -0500 The subject pretty much says it all. Does Asterisk support DNIS, and if so, what kind of connection is required? (T1, PRI) I've got a wink start T1. I've read comments that say the DNIS will be seen as an extension, but I'm seeing each digit of the DNIS as a separate extension. So in my case I send DNIS of 12345, Asterisk will jump from extension 1 to extension 2 to extension 3 to extension 4 to extension 5. Only executing the first one or two lines in each. This is a PITA! And make absolutely no sense to me. Matt already replied to your other posting of similar content. I'm also a bit confused. Do you mean you have observed that Asterisk is brought into the intended context, but start to react to digits in DNIS one after another? If so, can you estimate the interval Asterisk stays in each extension? If this is true, it seems to suggest that your provider is sending DNIS as a DTMF string after Asterisk has answered the call. Isn't this a bit weird? What does the card's manual say about DNIS (with wink start)? Yuan Liu Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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
[asterisk-users] How to configure Asterisk queue with Vonage account?
In http://www.voip-info.org/wiki-Asterisk+agents as followings, what type of channel of 28 and 29 is? agents.conf [agents] agent = 1001,4321,Wayne Kerr queues.conf [queue1] member = Agent/1001 extensions.conf exten = 28,1,AgentLogin(1001) exten = 29,1,Queue(queue1) I use the following in extension.conf with Vonage softphone account, it works well to call SIP extension 1001. exten = 180xx,1,Dial(SIP/1001,20) If using agent to login a queue, how to transfer the call to the queue first? I try two approaches, no one work. 1. exten = 180xx,1,Dial(SIP/28,20): no SIP 28 extension registered in Asterisk. 2. exten = 180xx,1,AgentLogin(1001) exten = 180xx,2,Queue(queue1): when calling that number, agent login. after hangup, agent logoff. How to keep this agent login in the queue. ___ --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] F1000 web configure
I have 8 - F1000G utstar phones. on a couple of them I can configure them by the WEB interface with no problem. On a couple of the them I cannot. I get no response when I point the browser to them. THe units work. keypad is fine. I can examine the network values - specifically the DHCP address. etc... Just cant get the web interface to do anything. Has anyone had this issue and know how to enable the WEB interface? Thanks so much, Jerry ___ --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] MixMonitor RingBack Tone Issue
Jean-Marc Salsa wrote: exten = s,n,Dial(SIP/[EMAIL PROTECTED],30,r mailto:SIP/[EMAIL PROTECTED],30,r) Everything works perfectly, except when the softswitch, or the PSTN sends back RingBack Tone. I can see the RTP flow arriving to Asterisk, but, it seems that Asterisk doesn't forward it to the other party (next-hop). Yes because you have the r in there, asterisk sends its own ringing. If you want ringing to be heard from the PSTN, you need to leave that option disabled. ___ --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] End Wrap-up Time?
I did it really only for our use. Because we manage our queue members solely through the manager interface, the implementation only works by issuing a command while connected to the manager port. The patch also adds 'Wrapuptime' as a return value to a queuestatus on the management port and changes the manager interface to not log every command received to the debug log unless the debug option is set. The diff can be found at http://www.omnis.com/queueendwait.diff. Matt wrote: Where is this patch? On 2/16/07, *James Fromm* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I patched 1.4.0 to add a command to the manager api in the queue application to implement the end wrap-up time I was asking about. All the command does is modify the 'lastcall' timestamp for the queue member by subtracting the value of the queue's defined wrapup time. Andrew Kohlsmith wrote: On 2/15/07, Andrew Kohlsmith [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: This is coming right out of left field, as I've never set up an Asterisk queue or agent system, but is it possible to pause and unpause while in the wrap-up time? What happens? Does the wrapup time go away then? Might be a counter-intuitive way around it if so... On Thursday 15 February 2007 4:34 pm, Matt wrote: I tried that. It didn't work :( What if we patched Asterisk to do just that? What could the repercussions be? They're already pausing/unpausing, so having the wrapup time auto-zero on unpause seems a non-issue... -A. ___ --Bandwidth and Colocation provided by Easynews.com http://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 http://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: FW: [asterisk-users] Problem Transferring Direct to Voicemail
Savoy, Kevin - Williston, ND wrote: Well thanks to those who did reply. I guess I'll have to live with it until somehow it gets fixed. The reason I upgraded to 1.4 is that there were three or four other issues I had that this fixed. Going back just isn't really an option since those issues were bigger then this one. Guess we'll live with it for now. I strongly encourage you to file a bug, as the developers need feedback to make improvements on 1.4. -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
[asterisk-users] Fwd: Can anyone help me out with Polycom 2.1 firmware please?
Any kind Polycom dealers out there? -- Forwarded message -- From: Eric Bishop [EMAIL PROTECTED] Date: Feb 14, 2007 8:10 PM Subject: Can anyone help me out with Polycom 2.1 firmware please? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Would be greatly appreciated ___ --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] Fwd: Can anyone help me out with Polycom 2.1 firmware please?
I can provide Polycom phones, and I have provisioning scripts. Is that what you need? Eric Bishop wrote: Any kind Polycom dealers out there? ___ --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 callerID
Hi Allen All, I had posted this kind of problem 2 weeks ago but seems nobody from here encountered yet. So I haven't received any reaction as of the moment. The problem with AudioCodes' FXO is that I cannot make it work without defining endpoints number. Once a number is defined, this number will serve as the callerID or will be displayed from a call coming from the FXO/PSTN. I guess same thing with FXO cards installed directly to an Asterisk Server. I have not find the solution yet until this time. Hope somebody from AudioCodes could share solutions on this matter. Allen Casteran [EMAIL PROTECTED] wrote: voip crazy wrote: Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I make a call directly using one of this line, the callerID is sending correctly. With the same zapata config file and the Freepbx 2.1.3, the callerId was sending correctly. Any clue will be welcome Again, your description is not clear. If your problem is that calls coming in to asterisk are not displaying caller ID on your phones, then you need to make sure that your CO lines are configured by the carrier to deliver caller ID. As a test connect a basic analog phone that has caller ID capability and call the line. If your simple phone displays the CallerID that you are calling from your line supports it and Asterisk should pick it up. If you do not see the caller ID on the analog phone when directly connected to the CO line, then call your carrier and ask them to provide Caller ID on your lines. I had this exact situation this morning, so yes it happens. If your problem is calling OUT from asterisk and your caller ID not getting displayed on the phone you are calling, that is also a function of the carrier and something you have NO control over. You should see something on the far end even if its Private unavailable or blocked. Call your carriers and ask them to check their set up for your phone lines. Allen. ___ --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 - Food fight? Enjoy some healthy debate in the Yahoo! Answers Food Drink QA.___ --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] X100P ring detection failure
My home * system I use for test/dev stuff has recently started to miss all calls. I have two of the X100P boards from x100p.com that are about a year old in it. They both have worked in the past no problem. At some point in the past couple of weeks they both stopped answering the phone. I don't have a specific date because only I and a couple of other people call in on those lines. Dialing out on the channels works just fine. I took the opportunity to reinstall 1.4.0 (zaptel and asterisk) to see if maybe it was a software issue, and I also swapped around the analog lines and made sure that they were still ringing if plugged into other equipment. I checked with ztmonitor and you can see the RX level go up when the ring comes in, but still no detection in *. I don't get the Starting simple switch message or anything like that anymore. I blew away my zapata.conf file and tried an example from the mailing list with no success. I tried turning on distinctive ring detection to see if maybe a subtle change in cadence was causing the problem but it did not detect anything either. I can provide config files upon request, I just don't want to clutter up the mailing list with them if it turns out it's not needed. Any suggestions would be wonderful. I am planning on buying an TDM400P based card but for the time being I'm unemployed and would like to avoid the expense. Also if my problem turns out to be software based I'd rather not have shelled out the money on hardware to fix it. Thanks -Scott ___ --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] IAX vs SIP - Getting Asterisk out of the media path
If a call comes into my Asterisk server on a DiD provided by an ITSP and the dialplan sends that call to another external number throught the same ITSP's network, I don't want the RTP packets to pass through my server once the call is bridged. I have had great success getting this to work using IAX, but I have not been able to get this to work with SIP. The call is bridged OK (media at both ends) but the media continues passing through my network. The default behaviour for the Dial command is to have Asterisk step out of the media path provided you avoid some options like tT, which I do, so this should work. One interesting note: In an Ethereal trace, I see 407 Proxy Authentication required just after the INVITE to the callee. Could that be part of the problem? If so what's the fix? I thought it had something to do with the auth parameter. I am: - Behind a NAT, - Running Red Hat 9.0 - Running Asterisk 1.2.14 How do I stop the media passsing through my Asterisk server after a call between two external parties has been bridged? My sip.conf and the dial command I use are below. Thanks, Hugh ;*** Dial Command *** exten = _6136930630,n,Dial(SIP/[EMAIL PROTECTED]) ; SIP.conf ** [general] ; context=incoming-bogus-calls bindport=5060 bindaddr=0.0.0.0 maxexpirey=3600 defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; externip=999.99.999.99 ;Outside address localnet=192.168.0.148/255.255.255.0 ;Inside Network ; register=6135551234:[EMAIL PROTECTED]/6135551234 ; [6135551234] type=peer ;auth=md5 auth=6135551234:[EMAIL PROTECTED] username=6135551234 fromuser=6135551234 fromdomain=myITSP.ca secret= host=sip02.myITSP.ca port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 insecure=very context=incoming-sip -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.411 / Virus Database: 268.17.36/681 - Release Date: 11/02/2007 ___ --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] Distinct call permissions for each user
someone please give me one example?
[locals]
exten = _NXX,1,Macro(outcall,${EXTEN})
[longdistance]
exten = _1NXXNXX,1,Macro(outcall,${EXTEN})
[macro-outcall]
exten = s,1,Dial(SIP/[EMAIL PROTECTED])
exten = s,2,Dial(Zap/.../${ARG1})
[fullaccess]
include = locals
include = longdistance
include = ...
[restricted]
include = locals
include = ...
Put user A into the restricted context, and user B into the fullaccess
context. You can include other extension (i.e. services) and implement
roll-over onto a backup trunks in macro-outcall.
You can of course also simply it and only have two contexts and no macro, etc.
--Luki
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[asterisk-users] manager command queue...
...I am having trouble deciphering the returned status line, it seems to return 1-5 as far as I can tell. i am only aware of the status codes produced by ExtensionState, which does not return a 5. I cannot figure out why the codes are diffferent. Can anyone help? Or map the codes for me, i have googled my eyeballs out of the sockets. ___ --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 with busydetect and cell phones
I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal correctly and clears the line. If I call from an internal extension to a cell phone and then hangup the cell phone Asterisk will never detect the busy signal though it is clearly there. Asterisk will happily sit there listening to the busy signal. I suspect that the busy signal styles are slightly different though it is undetectable to me. How can I fix this??? It causes severe issues when a call is forwarded to a cell phone via the Zap interfaces as once you hangup the cell phone Asterisk never releases the channel. zaptel.conf loadzone=us defaultzone=us zapata.conf [channels] language=en ; include zap extensions defined in AMP #include zapata_additional.conf ; TDM Port #3,4 plugged into PSTN ;AMPLABEL:Zap Channel %c context=from-zaptel signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes rxgain=0.0 txgain=0.0 immediate=no busydetect=yes busycount=4 ;busypattern=500,500 ;answeronpolarityswitch=yes ;hanguponpolarityswitch=yes ;callprogress=yes ;progzone=us channel = 3-4 _ Mortgage rates as low as 4.625% - Refinance $150,000 loan for $579 a month. Intro*Terms http://www.NexTag.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
