Re: [Asterisk-Users] Polycom dialplan restriction
As an example, here is my custom digitmap: digitmap dialplan.digitmap=9,911|9,411|0T|00T|1xx|9,011x.T|9,1[2-9]xx[2-9]xx|9,[2-9]xx|*7x|7x|*1xx|*8 The | is used to separate different entries. The comma means that it'll keep providing the dial tone after hitting 9. If you see a T after one of the entries, that means it'll wait till the timeout expires, if theres no T, then it'll dial that number as soon as you hit a match. Extensions in this office are 1xx. The reason i have 7x is for the park extensions. *8 is for call pickup (pickupgroup, callgroup) *7x is for various things such as call waiting, etc 0T and 00T - i don't think I even have dialplans setup for these... The rest are pretty self explanatory, for local, long distance, and international dialing. The international dialing has a T at the end, because different countries will have different numbers of digits. Any questions, let me know. Roman Carlos Chavez wrote: I am having a problem with some Polycom 601 phones. If I dial without picking up the handset or selecting the speaker I can dial numbers that are any lenght. But if I pick up the handset or are using the speaker I can only dial numbers that are 8 digits. When I dial the 8th digit it dials immediately. Obviously this creates problems when I am dialing long distance numbers or anything that needs more than 8 digits. Is there any way to increase the number of digits before the number is diales automatically? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth 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] Static problems with Asterisk + Polycom phones
Hey all, I'm having problems where there is significant static when making SIP - PSTN calls. SIP - SIP and SIP - VM calls are totally clear and fine. Here's the setup: Polycom 601,501, and ten 301s. Digum 2400 TDM card w/echo cancelling, 12 FXO ports. The TDM card is on IRQ 5 with nothing else on it. Server Specs: Asus P4P800E Deluxe P4 3.0 Ghz 1 GB Ram 80 GB SATA HD - There is no static when using a normal phone direct to the 66 block. - The sound is also a bit low, and bumping the volume on the Polycom phones makes the static alot worse (obviously) zapata.conf settings: [channels] language=en context=from-pstn signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=128 echocancelwhenbridged=yes echotraining=500 rxgain=6 txgain=3 group=0 callgroup=1 pickupgroup=1 immediate=yes faxdetect=no zaptel.conf settings: fxsks=17-24 loadzone= us defaultzone = us - Running Fedora Core 4 - Kernel 2.6.14-1.1653_FC4smp - USB is completely disabled. cat /proc/interrupts: CPU0 0:3115334 XT-PIC timer 1: 8 XT-PIC i8042 2: 0 XT-PIC cascade 5: 12453626 XT-PIC wctdm24xxp 8: 1 XT-PIC rtc 10: 93751 XT-PIC libata 11: 907892 XT-PIC SysKonnect SK-98xx, eth1 15: 111542 XT-PIC ide1 NMI: 0 LOC:3115228 ERR: 0 MIS: 0 Any other information you need to help me figure this out, please let me know. - Roman ___ --Bandwidth 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] RJ21-RJ11
Ing. Germán González B. wrote: Hi!! I'm looking for an adapter RJ21 to 24 RJ11 for a TDM2400. Somebody can help me with some sugestions? Thks!!! --- Germán González http://leon.podernet.com.mx --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://shop3.outpost.com/product/1729164?site=sr:SEARCH:MAIN_RSLT_PG -- Roman Volf Keystreams Internet Solutions [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] Pri Gateway Hardware
Johnathan Falk wrote: Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesnt have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost and the quality? Thanks Johnathan Falk Network Administrator Clinton Community Schools Have you looked at vegastream? I've heard really good things about them: http://www.vegastream.com/vega400.asp -- Roman Volf Keystreams Internet Solutions [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] Immediate routing on 0 (DNIS)?
Ken D'Ambrosio wrote: I've got a T1 (EM wink). Our four-digit inbound DNIS numbers are in the range of 0600 - 1699. However, the second that the 0 is seen on an in-bound 06xx call, it stops listening for any more digits, and immediately tries to route the call. My 16xx numbers wait for all four digits before trying to route. Is there something, somewhere, that tells it to do an immediate route on seeing 0? I don't have much of anything in my extensions.conf file. I'm seeing what's going on via tail -f /var/log/asterisk/full Any suggestions? Thanks! -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Post your extensions.conf and what's on the CLI (asterisk -r) -- Roman Volf Keystreams Internet Solutions [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] Re: Asterisk Christmas Help request
5) How do I change the time zone for Asterisk? Currently the system time is correct but when I dial *60 it reports a different time (out by many hours). I'm not familiar with this option. Can you please tell me more or send me some link. FYI, this is the relevant extensions_custom.conf entry on an AAH system: exten = *60,1,Answer exten = *60,2,Playback(at-tone-time-exactly) exten = *60,3,SayUnixTime(,,IMp) exten = *60,4,Playback(beep) exten = *60,5,Hangup [Description] SayUnixTime([unixtime][|[timezone][|format]]) unixtime: time, in seconds since Jan 1, 1970. May be negative. defaults to now. timezone: timezone, see /usr/share/zoneinfo for a list. defaults to machine default. format: a format the time is to be said in. See voicemail.conf. defaults to "ABdY 'digits/at' IMp" -- Roman Volf Keystreams Internet Solutions [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] Asterisk Hardware recomendation
Krystian Filiks wrote: What about plain g729? My main concern is the Hardware, anyone that can tell me if this Supermicro 6014H-32 is stable and sutible for asterisk? Supermicro Superservers are traditionally extremely stable and reliable. -- Roman Volf Keystreams Internet Solutions [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] MySQL Support For OS X
Michael Stearne wrote: Does anyone have the MySQL add-on as a binary for OS X? Or am I getting it wrong and MySQL is installed by default? Thanks. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Did you actually try to search for it, or did you just blindly post? http://dev.mysql.com/downloads/mysql/4.1.html Scroll down to where it says Mac OS X downloads. -- Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk starting problem
Bharat M. Sarvan wrote: Hello Everybody, I am having problems with starting Asterisk. The message what I am getting is; May 11 15:41:32 WARNING[5031]: res_musiconhold.c:728 moh_scan_files: Cannot open [cdr_addon_mysql.so]May 11 15:41:32 WARNING[5031]: loader.c:305 __load_resource: libmysqlclient.so.10: cannot open shared object file: No such file or directory May 11 15:41:32 WARNING[5031]: loader.c:463 load_modules: Loading module cdr_addon_mysql.so failed! I have configured the modules.conf for loading the cdr_addon_mysql.so. But still the problem persists. If you could please help me to figure as to whats wrong, it would be very kind of you. Regards, */Bharat M. Sarvan/* ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sounds like you are missing the mysql client libraries. -- Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Warning of the Asterisk server
Yao, Yuanbin wrote: Hi, I am trying to hook up Avaya 4602 SIP phone to Asterisk server, but got the following warning: May 10 15:45:15 WARNING[2042]: Unexpected bind error: Cannot assign requested address May 10 15:45:15 WARNING[2042]: Unable to create RTP session: Cannot assign requested address May 10 15:45:15 WARNING[2042]: Unable to build sip pvt data for MWI Can somebody tell me what went wrong with the Asterisk configuration? Regards, yyao ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Please stop double posting your questions. This will not help you get any answers. -- Roman Volf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blind Transfers - any ideas?
Have you looked here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] Jason Brown wrote: Here is something I wasnt quite expecting from a business deployment, and dont have an answer for. Maybe one of you do. Incoming call comes in. Rings 5 times to receptionist, then goes to menu system. So the receptionist answers a call, and blind transfers the call to extension X. The poor sap at extension X sees a caller ID Displayed on his phone of receptionist. He picks up the phone and says Hey Michelle wassupProblem is that it isnt Michelle. Its the CEO calling from Wisconsin. This is a problem. Its any transfersI am using blind transfer as the example. After someone picks up the call, how do you make sure that proper caller ID info is passed down the line? If its a real extension/extension Call, I want the extension as caller id, but if its a blind transfer of an inbound call, I want to pass the real callerid info so they know its a real Phone call. I am using Polycom phones with fresh CVS of asterisk. Thanks Jason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hitachi WIP-5000/IP-5000 firmware
Have you tried putting both access points on the same channel? Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] Jim Meehan wrote: I've got a Hitachi WIP-5000 phone. Seems to work well with my Asterisk setup, except for a few annoyances: 1) If the phone has been sitting unused for a while, and I dial an outbound call, it often fails. Doesn't matter what number I'm calling. If I redial, it always goes through fine, and all subsequent calls also go through fine, until it's been sitting around idle for a while again. Incoming calls always come through, regardless of how long it's been sitting. Maybe a NAT problem? I haven't started looking through SIP packet logs yet, but that's my next step. 2) I've got two 802.11b access points in my house, same SSID, one on channel 1, another on channel 6. The phone seems to stay associated with the access point that it first registered on, unless I do restart network on the phone. Even when the other access point is much closer with a much better signal. Both my Windows and Mac OS laptops switch between APs at will, depending on which is stronger -- seems like the phone should do the same. Anyone else noticed these issues? Also, I've got firmware v1.5.2 on my phone. Was trying to find a link to see if there's anything more recent. Anyone have a newer version or know where to get one? Thanks, Jim Meehan Oakland, CA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Google Group?
I setup this google group because Google seemed to be good at threading the topics from the list. I have noticed that many threads don't go as well as planned and wind up in the wrong place. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Google Group?
Oh and it was just a test to see how it worked. Pretty easy to setup Asterisk-users On Apr 8, 2005 8:47 PM, Roman Volf [EMAIL PROTECTED] wrote: I setup this google group because Google seemed to be good at threading the topics from the list. I have noticed that many threads don't go as well as planned and wind up in the wrong place. -- Roman Volf [EMAIL PROTECTED] [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set system time over the phone
Another way is to do:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (echo ${EXTEN} /tmp/datetime )
Then have a cron job that runs every minute to check if file exists. For
example:
#!/bin/bash
if [ -f /tmp/datetime ]
then
date `cat /tmp/datetime`
rm -f /tmp/datetime
fi
This should work fine.
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Matt Riddell wrote:
Peter Bowyer wrote:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
On the console Asterisk reports the command Dial 04021305 exits
non-zero.
You need 'Read' instead of 'Background'.
No, because his next line is _.,1 so it will actually use the extension.
His problem is just one of permissions. Maybe he should use a suid
prog to set the date.
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Re: [Asterisk-Users] Asterisk Newbie
Or if google is too complex, http://asterisk.keystreams.com Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] Robert Webb wrote: On Tue, 15 Mar 2005 11:56:18 -0500 Fabian Borot [EMAIL PROTECTED] wrote: Hello all I have been learning * from almost 1 month now. It looks really powerfull. I have some problem trying to find previous post, or solutions to common problems, advice to newbies etc in this mailing list. There is no a forum-like tool to search thru the posts by keyworks for example. Please correct me if I am wrong. Go to Google, in the search box type site:lists.digium.com without the quotes then type in what you want to search for. THis will limit all searching only to the Digium lists for asterisk. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SIP vs. IAX
Because SIP works with things other than Asterisk. IAX does not. Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] Joseph wrote: I'm just curious why Sipura isn't using free IAX protocol with their devices instead of SIP? With IAX NAT traversal would have been easier, so why are they using SIP. Is there any politics in it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to get 2 SIP phones to work
It would be helpful if you pasted the relevant sections of sip.conf and extensions.conf Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Im new to Astererisk. I compiled the latest CVS and setup the server. It looks like things are working. I'm running kphone, x-lite and sjphone to test things out. The kphone (local to the asterisk server) can call and receive calls from any of the 2 windows machines. The first windows phone I start I can send/receve calls the second one I cannot. I. No matter which one I start first only the first one works. The linux kphone can still call/receive from any of the 2 windows machine. I dont have another linux box to see if another kphone could send/receive. Everything seems to register fine in asterisks. The 2 windows machines are on seperate servers and in the same subnet. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending Voicemail's to two email addresses
In case you didn't get the last 5 responses, you just need to create an alias for the two email accounts. But honestly people, do you not read the rest of the thread before responding? Its already been answered. Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] C F wrote: yes, create an email address on you MTA that will deliver the message to 2 mailboxes (sometimes called a DL for distribution list) On Wed, 02 Mar 2005 14:32:30 -0500, Randy Johnson [EMAIL PROTECTED] wrote: Is there a way to send a voicemail to two different email addresses when a caller leaves a message? Thanks a bunch! Randy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Searchable Asterisk-users archive available
For those newbies who seem to not know google exists, I've setup a searchable forum located at http://asterisk.keystreams.com/ . Yes some of the threads are doubled up, but that sort of conversion is not perfect so just use the search feature. It currently has 2002,2003, 2004, and Jan/Feb of 2005. It will *not* be updated in real time (at least not for now) Please direct flames/questions/comments to [EMAIL PROTECTED] -- Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Administration manual for Sipura-841?
Have you seen the user guide? http://www.sipura.com/Documents/SPA841UserGuide.pdf Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] Scott Bussinger wrote: There isn't an Admin Guide for the SPA-841 as far as I know. However, I have found that the Admin Guides for their other products are VERY helpful. The firmware is very similar between tbem. I've got the documentation for the 2000/3000 units, but I was wondering about some of the features in the 841 like how the multiple lines/appearances configuration stuff works. So far I'm quite happy with the 841's. I only wish they had a little 10/100 ethernet hub in them to make hooking them up a little cleaner. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
