[asterisk-users] E1 information
Hi folks, I have a E1 interface and i need to get some informations about it, like: Informations about the Layer 2 (state - active, inactive) HDLC messages (time, channel, events ...) Messages from Layer 2 and 3 (Rec. Q.921 and Q.931) Number of sent messages and bytes, received messages and bytes Does asterisk has any support to those informations? I mean, Is it possible to get those informations from asterisk? Is possible to use the instance of libpri that asterisk uses? I found this info at the libpri code, but i need a way to get it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Free US Based Echo Test
Dear friends, As a community service, FailSafeVoip is providing a free US Based Echo Test. The service is running on a high performance asterisk box and is connected via a fully TDM T1-PRI. The test server is based in Michigan. The test extension is written simply as: s,1,Answer s,2,Echo s,3,Hangup Currently, The DID Is limited to 5 simultaneous connections. Please use this for anything you desire, but we do kindly ask that you not abuse the DID so that everyone may take advantage of it. DID Number: 586-408-9866 FailSafeVoip www.failsafevoip.com [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it oranother TDMoE bridge?
For high-availability Asterisk when using PRI Lines, you might also want to check out our product (FSV-4PFS). It's available at www.failsafevoip.com Bill Also to add to the last post does this device have hardware echo cancelation? if it does it could be a great replacement, if not may not be what I'd really want to use. Thanks, tom I've been looking at the RedFone foneBRIDGE2 2e1 product here: http://www.mapleleaf-technologies.com/webstore/redfone_fonebridge2_2e1.php Has anyone used this device (or something similar)? What were your thoughts on it? On the surface this seems like a perfect method of building high availability Asterisk environments, but I'm a little hesitant to spend a few grand just to find out it's a pipe dream. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ordering BRI From ATT
Hello everyone, I'm hoping someone can help me with this. I have a business customer in the U.S. (Michigan, ATT Territory). I need to get 4 trunks into an asterisk Box. My intention is to use an Eicon Diva Server card with 2 BRI Circuits. The reason for this is that the business needs DID's on the trunks (20 of them). A full or fractional PRI is overboard for them, as they will never need more than four channels. I also don't really want to go with any kind of analog trunk for other reasons (Disconnect supervision, potential echo problems, etc..) I've called ATT About a dozen times now, and no-one can tell me who to call in order to get a BRI. Is anyone familiar with how to order a BRI from ATT (Or a CLEC that covers Michigan)? I need 2 BRI Circuits (4 B Channels) with 20 DIDs accross the 4 circuits. Also, can anyone confirm that an Eicon Diva Server BRI Card will in fact support North American BRI? Are there any less expensive cards that will handle North American BRI? Any help highly appreciated Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip port= not working
Then there is no way to make asterisk listen on multiple port ? Currently iptables 5091 forward to 5060 is working but this should really be a asterisk feature :( On 16/12/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Mail list wrote: Yes i read that on voip-info wiki but i have bindport = under device (extension) which should make that extension work on other port but its not working . :( No, bindport= under the device section is ignored because it is not supported. ___ --Bandwidth 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] a2billing without IVR
Hello list, Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk). I want to dial the destination number to the asterisk. for example: user dials, exten =_011.,1,DeadAGI(a2billing) system will connect the destination and bill them. but right now we need to enter the destinationfollowed by the IVR prompts which i dont want. Thanks in advanved if anybody can help me. best regards shaon ___ --Bandwidth 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] anyway to a2billing without IVR
Hello list, Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk). I want to dial the destination number to the asterisk. for example: user dials, exten =_011.,1,DeadAGI(a2billing) system will connect the destination and bill them. but right now we need to enter the destinationfollowed by the IVR prompts which i dont want. Thanks in advanved if anybody can help me. best regards shaon ___ --Bandwidth 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] Group dial CDR
hello list, i have a dial plan exten =12345,1,Dial(sip/100sip/101sip/102,30,rt) so when calls come to 12345 all the phones 100, 101,102 rings. if any person receives the call it connects the call with no problem. but in the CDR it does not show who receied the call. it shows SIP/100SIP/101SIP/102 . how can get the information who received the call. thanks Best regards shaon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer caller ID
hello list, in my asterisk i have blind transfer and attendent transfer. when call Z which is a public call through Capi(BRI) is received by user A he can see the Caller ID of Z and if user A blind transfer the call to user B, user B can see the caller ID of user Z but when user A attendent tranfer the call to user B, user B does not get the caller ID of user Z. same timeit is notrecorded correctly in the CDR. how can solve this problem please help. best regards shaon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 box single Asterisk
hello list, ineed to setup an asterisk system with 5 ISDN trunks. ifound C4 cards but they are very expensive.i found that if i use 5 AVM Fritz! cards itwould bevery cheap. i want to use 2 boxes. 3 in boxA +2 in boxB=5 isdn. and i want,this two boxs to work as a single box so that one box can share ISDN hardware from other box. this system will be serving a call center. currenly we are using a panasonic PBX system but it is driving us crazy. we want to keep the existing pbx setup and add asterisk with it to handle the call center operations. we also need to communicate with pbx users from Asterisk. our pbx has6 analog trunks. so we can use TDM400P please help how can i solve this situation will low cost and performance. best regards shaon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Australian Dial tone TDM400P
hello asterisk users, i an using asterisk cvs 1.0.9 in a pIII 733mhz 256MB RAMredhat 9. i have a TDM400P with 2FXO and 2FXS modules. in my fxs i want to get australian dial tone and for all asterisk operation i want to use australian tones. by default it is US. to change this i have edited following files: /etc/zaptel.conf (loadzone = au, defaultzone= au) /etc/asterisk/indications.conf (country= au) but for all kind of signals and tones i stillget US tones. i restarted asterisk and run it #asterisk -vc no luck. please help best regards shaon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using E1 without power off simence pbx
hello everybody, i used asterisk with 2 ISDN BRI AVM cards in paralel with a panasonic ISDN pbx for testing putpose. is this also possible to use E1PRI to use in parallel with a simence PRI pbx for test purpose? |---asterisk public line| |---PBX best regards shaon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] res_features.so (Call Features Resource) not loading
hello everybody, i have updated my rpm asterisk to current cvs 1.0.9. I had been usingrpm asterisk which comeswith suse 9.2. the main reason i updated my asterisk to get the attended transfer feature. i haveinstalled anothercvs 1.0.9 asterisk in Redhat 9 and it works perfect. here what i found: in suse 9.2:*CLI show modules . res_features.so Call Parking Resource Note: it shows CallParking Resource . .. in Redhat 9: *CLI show modules.. res_features.so Call Features Resource Note: it shows Call Features Resource.. . i think asterisk is not loading call features resources in suse 9.2 so i also added a line in /etc/asterisk/modules.conf load =res_features.so . still attended tranfer feature is not working. iwant to use suse 9.2 as it got IDSN driversupport. i use2 BRI cards. please help Thanks in advance shaon (AU) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where to buy POLYCOM phones?
Try www.VOIPSupply.com they have the model 300, 500 and 600 phones available. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker Sent: Monday, October 18, 2004 5:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Where to buy POLYCOM phones? Jonathan Miller wrote: Hi all, I'm trying to put together a list of gear w/prices to implement an asterisk system. Does anyone know a good place to buy polycom phones? Their website isn't much help. Specifically looking for IP500 and IP600 phones. Thanks again! In Canada you can get them from CCP. West Canadian contact: Stacey Chamberland Canadian Communication Products Inc. 3657 Wayburne Drive Burnaby, BC V5G 3L1 [EMAIL PROTECTED] tel: 1-800-665-5726 fax: 1-604-263-9399 Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.voxbox.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please help I fear I have missed something very important! but what?
Sorry about this, I have been struggling with the basics of my
asterisk config.
I set up two sip peers and two phones. And I set up lots of dial
masks for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get incoming calls
to work. So I have gone back to a very basic FWD config, with one phone which
as far as I am aware should work, but doesnt. I cannot find info on how
to fix this.
Below is my sip.conf
[general]
port = 5060
bindaddr = xxx.xxx.xxx.xxx
context = sip
register = 2:[EMAIL PROTECTED]/1001
[fwd]
type=friend
secret=xx
username=xx
host=fwd.pulver.com
;
;
[1001]
type=friend
username=xx
host=dynamic
secret=xxx
callerid=Home
1001
dtmfmode=RFC2833
mailbox=1001
context=sip
and here is my
extensions.conf:
[general]
static=yes
writeprotect=no
;
[globals]
HOME=SIP/1001
;
[sip]
exten =
1001,2,Dial(SIP/1001,20,t)
include =
fwdnet
;
[fwdnet]
exten =
_8.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],t
Now as I said I can call out no probs by dialing 8 then the FWD
number, but incoming calls dont work, and as far as I can see that should
ring ext 1001 for 20 secs.
Could someone please help a complete Linux/Asterisk Newb, as
apart from this I have learnt a hell of a lot. But its
the last thing I need to solve.
The linux box for this testing has a
unfirewalled public IP address, so there is no problems with NAT
Please please can someone help. If I have missed something important then I aplogise, as I have been scouring the wiki
and the archives to no avail
Regards
Stuart Buchanan
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[Asterisk-Users] Pipecall problem
I have been a reseller subscriber of pipecall since they started, however I am really struggling to get pipecall to work for outbound or inbound calls. I get errors that the registration has timed out. I have tried many variations of the register command register = [EMAIL PROTECTED]/1000 register = sipx:[EMAIL PROTECTED]/1000 however none seem to work, the sip msg states it is unauthorised. A person called Tony Hoyle on this list managed to get pipecall to work for incoming as he said he could get the register command to authenticate two users i.e username and authuser, but he couldnt get it working for outbound. (Tony did you get it working?) Does anyone know any ideas as I am a bit stumped. This is what the section in my sip.conf looks like. [sipproxy.pipecall.com] type=peer secret= username=sipxx fromuser=0845xxx host=sipproxy.pipecall.com can anyone enlighten me to what settings to use. As I am head banging now Regards Stuart Buchanan -- This email is only for the intended recipient. If you have received this email in error please notify the sender and delete the message immediately. This email has been checked for viruses to ensure that any attachments are free from viruses. You should, however, carry out your own virus check before opening any attachment. We accept no liability for loss or damage caused by software viruses. --
[Asterisk-Users] WTS (200) Cisco ATA-186-I1
Condition: New Open Box Warranty: 90 Days Cost - $130/ea Minimum Order 5pcs Contact [EMAIL PROTECTED] for details Cory Andrews ++ b2 technologies 454 Sonwil Drive Buffalo, NY 14225 ++ email - [EMAIL PROTECTED] voice - 716.630.1555 X22 fax - 716.630.1548 web - www.ValueResale.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail main
How do you configure extensions.conf to let you punch out to VoicemailMain when an individual voicemail prompt has picked up? We have a few extensions set up. voice mail is extension 8500 and we have another extension for SIP on extension 12. SIP dials out fine. SIP can dial 8500 and get VoicemailMain. An incoming call is picked up by extension 12's voice mail if no answer. What if I call in on the incoming POTS line and extension 12's voice mail picks up. How can I have that voice mail switch me over to VoicemailMain if I press a number or * or # or something? Here's our config at the moment: exten = 8500,1,VoicemailMain exten = 8500,2,Hangup exten = s,1,Dial(SIP/12,10) exten = s,2,Voicemail,u12 Thanks - I've been looking for an example for this on the lists and google for a few days. Bill. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What failed here?
Could you have asterisk running and not allowing you to overwrite while trying to install? Do you have root rights to create files in the asterisk folders? John Chambers wrote: After doing cvs checkout -r v1-0_stable asterisk and typing the usual make clean ; make install, I got these messages: ... make[1]: Entering directory `/usr/src/asterisk/codecs' make -C gsm lib/libgsm.a make[2]: Entering directory `/usr/src/asterisk/codecs/gsm' /var/spool/asterisk -o src/k6opt.o src/k6opt.s make[2]: execvp: /var/spool/asterisk: Permission denied make[2]: *** [src/k6opt.o] Error 127 make[2]: Leaving directory `/usr/src/asterisk/codecs/gsm' make[1]: *** [gsm/lib/libgsm.a] Error 2 make[1]: Leaving directory `/usr/src/asterisk/codecs' make: *** [subdirs] Error 1 # It looks like something has gone badly wrong here, but I can't make any sense out of it. I dug around looking for 'k6opt*', and found them in codecs/gsm/src, and presumably that's what it was trying to compile, but this isn't much of a clue. There's no libgsm.a in any directory. From the look of the line starting with /var/spool/asterisk, I'd guess that some macro in some makefile has come up undefined, but that's not much info, either. Any idea what it might be trying to tell me? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP 3.0
Can anyone point me to where I might obtain the SIP 3.0 image for the ATA-186 Analog adapter. I'm willing to pay for it. I have a Cisco login but am apparently not authorized for this, just trying to get my fax working with asterisk and I need SIP 3.0. Any advise appreciate. Thanks Cory ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone working with NUFONE?
Curious if anyone has any feedback on Nufone voip pbx. Cory J Andrews ** b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 ** 866.44.B2TECH X22 local 716.630.1555 X22 fax 716.630.1548 *** [EMAIL PROTECTED] web http://www.ValueResale.com
[Asterisk-Users] Cisco AC Power Cubes for Sale
Wehave (2) cartons of (56) AC Power Cubes for the Cisco 7905, 7910, 7940 and 7960 IP Phones. These are brand new, and include the power cord. They come with a 1 year warranty. Cost is $17/ea, minimum order of 10 pcs. Cory Andrews***b2 Technologies454 Sonwill DriveBuffalo, NY 14225***voice - 716.630.1555fax - 716.630.1548email - [EMAIL PROTECTED]
[Asterisk-Users] WTS (200) AC Power Adapters for Cisco 7910 / 7940 / 7960 IP Phones
Have (200) Brand New power cubes (AC Power Adapter with AC Cord) - compatible with Cisco CP-7910, CP-7940, CP-7960 and equivalent "G" models. $25/ea - Minimum Purchase (10) Units. Email [EMAIL PROTECTED] if interested. Regards Cory Andrews***b2 Technologies*** web - www.ValueResale.com email - [EMAIL PROTECTED]
[Asterisk-Users] For Sale - (10) Dialogic D/240SC-T1 REV2
We have (10) Dialogic D/240SC-T1 REV 2 voice boards. Prefer single buyer, take all (10) for $675/ea Cory Andrews * b2 Technologies 454 Sonwill Drive Buffalo, NY 14225 * voice: 866-44-B2TECH X22 fax: 716.630.1548 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (3) Cisco NM-HDV-2T1-48 for Sale
Never used in production - $3750/ea Email [EMAIL PROTECTED] if interested. Cory Andrews * b2 Technologies * voice: 866-44-B2TECH X22 fax: 716.630.1548 email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: Restoring Cisco 7960 to defaults
Can anyone point me to some online documentation showing how to reset a CP-7960 to factory default settings. I have some that are configured for Callmanager and I want to get them back to generic default config. Any info is appreciated. Thanks Cory Andrews ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: Anyone looking for IP Phones?
My company has approx. 500 Cisco CP-7960G IP Phones that are coming out of service. They were deployed for about 6 months. These include the AC power adapter and station license. We also have some other related equipment. If someone is reading this and is interested, shoot me an email [EMAIL PROTECTED] Thanks Cory Andrews * b2 Technologies ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
