Re: [Asterisk-Users] What is asterisk
bmw suzuki wrote: Hello all ... mY first ever post in here. I am bit or (full) confused on what this program does.is http://does.is it useful if i have a alcatel pabx system.And i can bill my guests for their call charges etc.. can i use it on calling another computer on the network via Ethernet card.I have already read the Documentation,But if any one could clear me up on the above things. how can i call a regular PSTN landline phone Via this software through internet?Do i need dedicated hardware for this or an ethernet card will do. Some helpful references: http://www.amazon.com/gp/search/104-2683497-5764764?search-alias=apskeywords=asterisk ___ --Bandwidth 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
R: [Asterisk-Users] Capturing DTMF during a call
Thanks Kristian, but i just answered to call, how can i use the Read application? Thanks Giordano Grandis -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Kristian Kielhofner Inviato: lunedì 6 marzo 2006 18.15 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Capturing DTMF during a call Giordano Grandis wrote: Hi all, I have a simple and maybe also stupid question: if i'm in coversation on a Zap channel and the remote party send me a DTMF, could I capture it? Thanks all *Giordano * show application Read -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk add-ons - H323
How to upgrade h323 from Asterisk add-ons (from version 1.2.1 to 1.2.2)? In INSTALL they don't say anything about upgrade... Thank you for your time! -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth 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] Gmane - Asterisk Users Mailing List
Hi group! Does anybody knows about any news server that works the same way that Gmane www.gmane.com/ does it? I was satisfied with Gmane for few months, but now it seams that it doesn't work any more (no new posts in past few days). Now I'm looking for alternative. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth 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] Hangup issues
Hello, Load wctdm with debug=1, i.e, add this line to /etc/modprobe.conf: options wctdm debug=1 Then watch /var/log/messages (tail -f /var/log/messages will do it), and check when you are getting the first polarity reversal, you should get it before the first RING. If it happens that you get it when asterisk answers, that would explain your problem. BTW, is it a pstn line? or a gsm fct? If the later, you need to set it up for proper hangup detection in asterisk. Julian J. M. On 3/7/06, Carlos Prieto [EMAIL PROTECTED] wrote: Hi ! I have some issues, i don't know exactly if it's a busy detection issue. When i dial into the Asterisk box, and if i hang up before the Asterisk answers with the IVR Welcome message, the Asterisk goes on with the call. But, if i wait for the Asterisk to answer, and if i hang up, the Asterisk hangs up too. I have this parameters on zapata.conf: busydetect=no answeronpolarityswitch=yes hanguponpolarityswitch=yes callprogress=no I've tested with different values por busydetect set t yes and several busycount values. I'm using Asterisk 1.2.4 and Zaptel 1.2.3 with a Digium TDM400P with 2 FXO modules and Kewl Start Signalling. Thanks in advance for the help. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to receive faxes with SPA3000 and Asterisk setup
On Tue, 7 Mar 2006, Zach A wrote: I have SPA3000 receiving PSTN calls and also have a SIP line on the same Asterisk server with 5 extensions. Now there is a fax too which comes through the PSTN line. Fax calls have short rings. Can Asterisk somehow detect those short rings and send the call to the fax machine on one extension? Or is there any other way of receiving faxes. Can SPA3000 send fax calls directly to the fax machine when detecting the short rings? I need some solution to receive faxes. I use NVDetect to detect fax calls on the SPA3000 FXO, and then forward the call to a fax machine hanging on the FXS port. Works good for me. -Dan ___ --Bandwidth 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] IPv6
Hans Witvliet wrote: Can anyone inform me if voip can be used on a IPv6 network? Does any hard phones/soft phones/Asterisk support it? Google told me that there was/is a bounty on it, but that expired august last year. Furthermore, there used to be a patch (Bernhard Schmidt), but that one is about a year old. I presume it can't be used on recent versions of * Hans Hans VOIP can certainly be used with IPv6. Asterisk does not support it, but there are phones and softswitches that do. Regards -- Chris Hills | Tel: +44 (0)1527 572754 IT Services | Fax: +44 (0)1527 572901 North East Worcestershire College | Web: http://www.ne-worcs.ac.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] Periodic-announce in queues
Hei. I have a question about how to get the periodic-announce to work within my queues. I got the following test: extensions.conf -- exten = s,2,Queue(test|rtT|||200) Queues.conf -- [testqueues] strategy = ringall context = testcontext timeout = 250 periodic-announce-frequency=60 periodic-announce = queue-periodic-announce member = SIP/591 Log from show queues testhas 1 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2, SL:0.0% within 0s Members: SIP/591 (Unknown) has taken no calls yet Callers: 1. SIP/192.168.234.11-081afc30 (wait: 1:04, prio: 0) ::: After 60 sec it is still ringing, and the periodic-announce is not announced. Is this periodic-announce only announced when all agents are busy, or should it announced every 60 sec as I want it to do? Anyone knows where my settings are wrong? Regards, Fredrik Jensen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr records on transfer
On Mon, 6 Mar 2006 18:53:30 +0100 (CET) Christian Benke [EMAIL PROTECTED] wrote: Hello! i'm trying to set up transfer without using the respective asterisk-function but with the built-in phone functions. my goal is to have the first callleg billed to the caller and the second callleg to the callee, who is responsible for the forward(and i can't bill a unknown caller anyways) so far it's working without problems, but my cdr's are messed. with the help of the RDNIS-variable i've been able to set seperate records for each call-leg with the correct accountcodes, but the billsec are still written to the first callleg, the second callleg(originated by callee) receives 0 billsec, which is not what i want. the callee(the one who forwards the call), should be billed. since the local-channel is passed to the originating channel, it is clear that the billsec are added to the callers record. but is there any way to influence this??? since the phones have this functionality built-in, why should i ask my clients to use some keycombination to transfer calls and prevent transfer-by-button? As far as i've understood, the /n-option for the local-channel would do the behaviour i want - but how could i add it on a moved temporarily? kind regards christian can i assume this is a known problem? Can anyone at least confirm it? Or is my report unclear? I really appreciate any comments, this is a huge problem for me as my whole concept depends on it! Thanks!!! Chris ___ --Bandwidth 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] Periodic-announce in queues
Try this ...extensions.conf --; Queue with Music on holdexten = s,2,Queue(test|mtT|||200) ___ --Bandwidth 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 compiling ztdummy on centos 4, 2.6 kernel
On Mon, 2006-03-06 at 10:54 -0500, Sina Bahram wrote: Here is the compilation process of zaptel I did edit the makefile and uncommented the #ztdummy, although, after I did that, I get the make error of ztdummy being defined more than once. [snip] You don't need to uncomment ztdummy in the Makefile because if you are using a 2.6 kernel it will be built automagically. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ON DEMAND call Recording
Hi i have configured extensions to record voice conversions ON DEMAND on my [EMAIL PROTECTED] so how will we start the recording when the call is in progress.thanksGiridhar Bandi ___ --Bandwidth 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 Prepaid Card
Hi group, I am currently looking for a prepaid application that can do the following: Use the Caller ID/Card Number for authentication Can map a rate plan on a specific Caller ID/Card Number Supports prepaid functionality in terms of trunk connection. These functionalities seems feasible in A2billing but the problem is I cannot find a proper documentation of setting it up. Can anyone show point to the right direction? Does any one has a better suggestion? Thank you very much in advance! Leonimar Cape __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Problem compiling ztdummy on centos 4, 2.6 kernel
However, as I pointed out in my email, that doesn't make any difference. If I leave it commented ... I get the exact same thing: just minus the make file error Same behavior, same error messages with the /etc scripts, the modprobe's and with everything else. Take care, Sina -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Tuesday, March 07, 2006 4:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel On Mon, 2006-03-06 at 10:54 -0500, Sina Bahram wrote: Here is the compilation process of zaptel I did edit the makefile and uncommented the #ztdummy, although, after I did that, I get the make error of ztdummy being defined more than once. [snip] You don't need to uncomment ztdummy in the Makefile because if you are using a 2.6 kernel it will be built automagically. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED] and H323
Hello I attempt installing H323 at my [EMAIL PROTECTED] for this use asteriskathome-h323-1.0.zip but have next problem chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_show_channels': Please help for resolve this problem Viktor Tatianin ___ --Bandwidth 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] Meetme Participant Announcement
On 03/07/06 01:14 Douglas Garstang said the following: Hi Doug. I worked it out. I had commented out chan_zap.so in modules.conf as I didn't think I needed it. It was doing weird stuff, including not playing the participants joining. Weird. MeetMe needs a timing device to work correctly. you can either provide this thru a zaptel card or the ztdummy pseudo timer. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth 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] Bad Meetme() Bug
On 03/07/06 00:44 Douglas Garstang said the following: Anyone seen this? If not I guess I'll have to post it as a bug. Extensions.conf has this: exten = 123,1,Meetme(|dMic|) I dial 123, and enter my conference number. Asterisk asks me to enter my name. At this point I hang up. If I type at the Asterisk console 'meetme list 12345' it shows that I am a participant in the conference evenhough I hung up. sounds like the recording of the name is not timing out when the phone is hung up. open up a bug on this at bugs.digium.com. also state what version of asterisk you're using. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth 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] Calls between Asterisk servers using SIP? What about IAX (got it working w/ IAX but I have questions)
Hi everyone, I just spend the last two hours trying to get two asterisk boxes to transfer calls between eachother using SIP. I dont know why but I *could not* get the calls to authenticate! I think I got everything setup. There was Server A and Server B. I was trying to place a call from a users registered on Server A to a user regsitered on Server B. I setup the registration info for Server A and even had Server A registering successfully to Server B. However, whenever I would hand off the calls from server A to Server B, it would *always* say it failed to authenticate (passwords did not match). Here was my setup: SERVER A: register = serga:[EMAIL PROTECTED] [to_80] username=serga type=friend secret=test host=216.152.244.81 disallow=all allow=ulaw user=phone usereqphone=yes canreinvite=yes regseconds=0 cancallforward=yes dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very trunk=yes SERVER B: [serga] type=friend username=serga trunk=yes notransfer=yes secret=test context=302 host=dynamic qualify=yes DIALPLAN ON SERVER A: exten = 302,1,Dial(SIP/to_80/[EMAIL PROTECTED],30,r) It always says authentication failed. However I always noticed it showed the user as [EMAIL PROTECTED] This is the extension of the phone I am calling from. It seems it is trying to authenticate the actual phone I am calling from on Server A, and not Server A itself. Was I doing something wrong? I tried doing this with IAX and within 5 minutes I had it all working!! I feel it was too easy :-) However, this brings up a big question.Is IAX very reliable for this? I've heard from people that I should not use IAX under any condition because it really is not very reliable/thourough/consistant...etc. I am trying to start a VOBB company and will obviosly need a reliable setup. I am thinking to have all phones register to the servers via SIP and maybe just have all the servers transfer calls between eachother via IAX. Does this sound like a correct setup? - Gabe ___ --Bandwidth 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] ON DEMAND call Recording
You can activate the on demand recording on the [EMAIL PROTECTED] by adding w and W in the asterisk dial command option. It is located on the general settings under setup. Either the caller or the called party can initiate the recording by pressing *1. Leonimar, --- Giridhar Bandi [EMAIL PROTECTED] wrote: Hi i have configured extensions to record voice conversions ON DEMAND on my [EMAIL PROTECTED] so how will we start the recording when the call is in progress. thanks Giridhar Bandi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] [EMAIL PROTECTED] and H323
Hi Viktor, What is the version of the asterisk you are using? You should use the right version of Openh323 and pwlib to be able to compile chan_oh323 successfully. Currently using asterisk 1.2.4 used openh323-Mimas_patch2-src-tar.gz and pwlib-Mimas_patch2-src-tar.gz for compiling chan_oh323. Hope this help. --- Viktor Tatianin [EMAIL PROTECTED] wrote: Hello I attempt installing H323 at my [EMAIL PROTECTED] for this use asteriskathome-h323-1.0.zip but have next problem chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_show_channels': Please help for resolve this problem Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] ENUM lookup issues with e164.org
On 3/6/06, Olle E Johansson [EMAIL PROTECTED] wrote: 7 mar 2006 kl. 02.45 skrev Scott Call: Since e164.org added DNC and ADDRESS records my enum configuration has failed. Using both the old EnumLookup app and the new ENUMLOOKUP function, the lookups have consistantly failed since e164.org added E2U +ADDRESS and E2U+DNC records.There's an open bug report in the bug tracker for this. Checking the bug tracker when you find problems is propably a good idea - then youcan confirm that you have it too./OI checked and while there are a few enum related bugs (a crash and one that looks like the query is not being sent at all) I could not find one that seemed specifically related to the errors I was seeing. Please let me know the issue # you were referring to so I can add my comments to it, as I don't want to open a duplicate if it's not applicable. 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] Re: ON DEMAND call Recording
ya i found it it *1 to start recording from the caller end thanksGiridhar Bandi On 3/7/06, Giridhar Bandi [EMAIL PROTECTED] wrote:Hi i have configured extensions to record voice conversions ON DEMAND on my [EMAIL PROTECTED] so how will we start the recording when the call is in progress.thanksGiridhar Bandi ___ --Bandwidth 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] Help! Connecting two Astersik via SIP channels
Hi everyone, I want to call from one Asterisk to another Asterisk via SIP, but i dn't know how. I have found out something in these links: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels but I don't understand them very well. At first, I tried simply doing this: In SIP Client: exten = _1.,1,Wait(1) exten = _1.,2,Dial(SIP/192.168.0.51:5060,20) (92.168.0.51 is the Asterisk server machine) And nothing on the server side. When calling on the client side I get: Executing Dial(Local/[EMAIL PROTECTED],2, SIP/192.168.0.51:5060|20) in new stack -- Called 192.168.0.51:5060 -- SIP/192.168.0.51:5060-c870 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) debian*CLI And the server side doesn't seem to realiza anything... meanwhile Can please anybody help me??? I am a newbie, and I don't know how to carry on with my job... :(( Thanks in advance, -- María ___ --Bandwidth 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] Help! Connecting two Astersik via SIP channels
2006/3/7, María Chóliz [EMAIL PROTECTED]: I want to call from one Asterisk to another Asterisk via SIP, but i dn't know how. If you are connecting one asterisk to another, I sugest you to use IAX2. It is better in some ways. And I sugest you to ensure you compiled the speex codec and use it if your connection is through internet. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to receive faxes with SPA3000 and Asterisk setup
2006/3/7, Zach A [EMAIL PROTECTED]: I have SPA3000 receiving PSTN calls and also have a SIP line on the same Asterisk server with 5 extensions. Now there is a fax too which comes through the PSTN line. Fax calls have short rings. Can Asterisk somehow detect those short rings and send the call to the fax machine on one extension? Or is there any other way of receiving faxes. Can SPA3000 send fax calls directly to the fax machine when detecting the short rings? I need some solution to receive faxes. Is the SPA what must detect the distinctive ringing (if it has this feature). Otherwise, you can use the internal fax detection of Asterisk, but if you activate it, Asterisk will answer the calls as soon it rings and hears for a fax carrier. If it is detected, Asterisk can transfer the call to the fax extension or use it's software fax. An interesting replacement for asterisk software fax is using iaxfax+hylafax. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls between Asterisk servers using SIP? What about IAX (got it working w/ IAX but I have questions)
Hello Gabriel, IMHO, using IAX between * servers is a good choice, I dont see any problem in it. Actually I used it for sometime and never encounter any issue, but i had max 5 concurrent connections. regards, Umair bari On 3/7/06, Gabriel Afana [EMAIL PROTECTED] wrote: Hi everyone, I just spend the last two hours trying to get two asterisk boxes totransfer calls between eachother using SIP.I dont know why but I *could not* get the calls to authenticate!I think I got everything setup. There was Server A and Server B.I was trying to place a call from ausers registered on Server A to a user regsitered on Server B.I setup the registration info for Server A and even had Server A registeringsuccessfully to Server B.However, whenever I would hand off the calls fromserver A to Server B, it would *always* say it failed to authenticate (passwords did not match).Here was my setup:SERVER A:register = serga:[EMAIL PROTECTED][to_80]username=sergatype=friendsecret=test host=216.152.244.81disallow=allallow=ulawuser=phoneusereqphone=yescanreinvite=yesregseconds=0cancallforward=yesdtmfmode=rfc2833disallow=allallow=ulaw insecure=verytrunk=yesSERVER B:[serga]type=friendusername=sergatrunk=yesnotransfer=yessecret=testcontext=302host=dynamicqualify=yesDIALPLAN ON SERVER A: exten = 302,1,Dial(SIP/to_80/[EMAIL PROTECTED],30,r)It always says authentication failed.However I always noticed it showedthe user as [EMAIL PROTECTED].This is the extension of the phone I am calling from.It seems it is trying to authenticate the actual phone I amcalling from on Server A, and not Server A itself.Was I doing somethingwrong?I tried doing this with IAX and within 5 minutes I had it all working!!I feel it was too easy :-) However, this brings up a big question.IsIAX very reliable for this?I've heard from people that I should not useIAX under any condition because it really is not veryreliable/thourough/consistant...etc.I am trying to start a VOBB company and will obviosly need a reliable setup.I am thinking to have all phonesregister to the servers via SIP and maybe just have all the servers transfercalls between eachother via IAX.Does this sound like a correct setup? - Gabe___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is asterisk
2006/3/7, bmw suzuki [EMAIL PROTECTED]: how can i call a regular PSTN landline phone Via this software through internet?Do i need dedicated hardware for this or an ethernet card will do. To call to PSTN lines there are some alternatives: 1) install FXO hardware in your asterisk server. This allows you to connect PSTN lines to your PBX. 2) use a FXO to sip converter like Sipura SPA3000. Then, asterisk can connect to the spa via sip protocol (tcp/ip) and receive and make regular phone calls. 3) use a sip (or iax) PSTN provider. It will allow you to place calls and obviously will bill you for it. It is the same case as 2 but you don't own the PSTN access. Some providers can assign you a PSTN number to receive incomming calls, other don't have this service. The atvantage of this is the low prices of international calls. NOTE: To use regular phones with your asterisk pbx, use FXS cards, FXS to sip adapters (like SPA2100) or sip phones. -- Alejandro Vargas ___ --Bandwidth 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] Two Asterisk server
Hi, I have two Asterisk server linked by a IAX2 trunk with two PRI+DID: Site1: XXX100-499 Site2: YYY100-499 (I masked real number with XXX and YYY) PSTN PRI1 --- Asterisk1 ...IAX2... Asterisk2 --- PSTN PRI2 Users: - keep their extension when moved between sites - can be reached from PSTN with both XXXext and YYYext In other words, dialplan is shared between servers. Actually, we have two Alcatel PBX 4400 working in this way. Can I do this with Asterisk? Thanks Mimmus ___ --Bandwidth 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 compiling ztdummy on centos 4, 2.6 kernel
On Tue, 2006-03-07 at 05:04 -0500, Sina Bahram wrote: However, as I pointed out in my email, that doesn't make any difference. If I leave it commented ... I get the exact same thing: just minus the make file error Same behavior, same error messages with the /etc scripts, the modprobe's and with everything else. Take care, Sina You could try the rpms at http://www.laimbock.com/asterisk/ Regards, Patrick ps please don't top post (put your answer *below* the posting). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Tuesday, March 07, 2006 4:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel On Mon, 2006-03-06 at 10:54 -0500, Sina Bahram wrote: Here is the compilation process of zaptel I did edit the makefile and uncommented the #ztdummy, although, after I did that, I get the make error of ztdummy being defined more than once. [snip] You don't need to uncomment ztdummy in the Makefile because if you are using a 2.6 kernel it will be built automagically. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Periodic-announce in queues
Well, with music I manage to get the periodic-announce to work, but only when the queues had no available members it will play the periodic announcement within the periodic-announce-frequency. To manipulate this I figured out that you can set the timeout within queues.conf to: Timeout = 60 Then when you dial this exten you will be in the queue for 60 sec, you go out from the queue, the announce-messages is played, and you are back into the queue again, I do not want to exit the queue and join the queue again, I would like to have a message play while dialing. This is an irritating problem with Eyebeam which will popup every time when someone joins the queue, I also have some irritating problem with the Cisco 7940 phone with SIP software, which also switch lines when a new call comes in (if you example are dialing out with line 2, you will automatic jump to line 1 when a new call comes in) And I found out that it is the same with regular announcement, it will also only play when there are no agents available. I would like to announce messages after 60 sec within the queue, regardless if the agents are available or not and without having the dialer jumping out of the queues. Does anyone know how to solve that? Regards Fredrik Jensen From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CC Jay Sent: 7. mars 2006 10:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Periodic-announce in queues Try this ... extensions.conf -- ; Queue with Music on hold exten = s,2,Queue(test|mtT|||200) ___ --Bandwidth 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] fax receive using TDM400P, with Tzafir, Anton, Cosmin, Colin...
Dear friends:I have seen Tzafir, Anton, Cosmin, Colin and other very interesting peopleworking very hard with the fax, almost at the point to write a book (I hope some day they will for all of us). I have been reading and saving all of those mails carefully to find the key to my needs. But their knowledge is too high for me.Can any of you explain me, send me a document or refer me to a book where I can find step by step (for a person like me who doesn´t know linux more than a couple of commands) how to make the faxwork?My [EMAIL PROTECTED] with tdm400p w/4 fxo ports, seems to negotiate...but I don´t know where to find the files. Before I used to go to my webmail in de AMP and see some of the files there, and when I opened them, all pages where whith nothing in.What I want is the [EMAIL PROTECTED] to receive my faxes and then send it to my email.Thanks.Yrving Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.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
RE: [Asterisk-Users] [EMAIL PROTECTED] and H323
Hi I use Asterisk 1.2.1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of leonimar cape Sent: Tuesday, March 07, 2006 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] and H323 Hi Viktor, What is the version of the asterisk you are using? You should use the right version of Openh323 and pwlib to be able to compile chan_oh323 successfully. Currently using asterisk 1.2.4 used openh323-Mimas_patch2-src-tar.gz and pwlib-Mimas_patch2-src-tar.gz for compiling chan_oh323. Hope this help. --- Viktor Tatianin [EMAIL PROTECTED] wrote: Hello I attempt installing H323 at my [EMAIL PROTECTED] for this use asteriskathome-h323-1.0.zip but have next problem chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_show_channels': Please help for resolve this problem Viktor Tatianin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom voice.gain.tx.analog.handset and asteriskecho
Wilson Pickett wrote: I use 3 which is the default on my 501's and 600's No echo here Actually, admin docs warn us NOT to change this value, but I am not in the US. I don't always have echo, but when there is echo it almost always goes away by lowering the level into the handset (or headset mic). I am in the US, but sill considering it. Watching ztmonitor when people talk, has the gauge pegged to the max on outgoing. Even with the Tellabs in place, we still get a slight echo. Maybe I'll knock it down to 2. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanSpy
Someone have good sound on ChanSpy with SIP channelsa at an Asterisk 1.2.4 ?Mine is cracking all the time.-- Adrià Vidal ___ --Bandwidth 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] webvmail problems
I have done my make webvmail, what else do I need to do? How do you get to the site? Any help would be appreciated. Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 1-800-666-2833 x299 (608) 783-7560 x299 ___ --Bandwidth 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] MixMonitor
Hi everybody, I have the same problem, but I have just upgraded to 1.2.5. The changelog says this problem is fixed in this version, but I don't think so. Asterisk is still crashing. []s Alex Robertson 2005/12/18, Mohammad Shokuie shokuie at hotmail.com: Hi there, Any one confronted a crash in asterisk when using mixmonitor app. When i'm using the mixmonitor app on a briged call as soon as the called party hangs up the call asterisk crashes and the process terminates with following error message : Segmentation fault. Ouch .. error while writing audion data :: broken pipe but when the calling party hangs up, everything is smooth. Anyone has any idea on this issue? TIA. M. Shokuie Nia ___ --Bandwidth 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 change Budgetone dialtone?
Good day! Is is possible to change dialtone (and other tones as well) in BT-102? pgpH067L1PT2h.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom voice.gain.tx.analog.handset andasteriskecho
|While I'm asking about the Polycom ip500, the answers for all phones |where mic/handset/headset levels are adjustable would be of interest to |many I'm sure. | |For the ip500, the default value for the handset seems to be |voice.gain.tx.analog.handset=3 I have a number of IP600s and 601s that I was experiencing occassional echo with. I recently upgraded them to firmware 1.6.5, and rather than using my existing sip.cfg/ipmid.cfg that had been around forever I started fresh with a completely stock 1.6.5 sip.cfg file. My echo issues have disappeared completely. With the 1.6.5 version of the Polycom firmware the default value for voice.gain.tx.analog.handset=12. The default value for voice.gain.tx.analog.headset=3. I suspect you should update the entire voice section of the file (if you're not ready to start from scratch) since it contains default values for AEC, AES, NS, AGC, RXEQ, and TXEQ. I have pasted just the gains section below in case anyone want to compare it to their current settings. gains voice.gain.rx.analog.handset=0 voice.gain.rx.analog.headset=0 voice.gain.rx.analog.chassis=0 voice.gain.rx.analog.chassis.IP_300=-6 voice.gain.rx.analog.chassis.IP_4000=3 voice.gain.rx.analog.chassis.IP_601=6 voice.gain.rx.analog.ringer=0 voice.gain.rx.analog.ringer.IP_300=-6 voice.gain.rx.analog.ringer.IP_4000=3 voice.gain.rx.analog.ringer.IP_601=6 voice.gain.rx.digital.handset=-15 voice.gain.rx.digital.headset=-21 voice.gain.rx.digital.chassis=0 voice.gain.rx.digital.chassis.IP_4000=0 voice.gain.rx.digital.chassis.IP_601=0 voice.gain.rx.digital.ringer=-21 voice.gain.rx.digital.ringer.IP_4000=-21 voice.gain.rx.digital.ringer.IP_601=-21 voice.gain.rx.analog.handset.sidetone=-14 voice.gain.rx.analog.headset.sidetone=-24 voice.gain.tx.analog.handset=12 voice.gain.tx.analog.headset=3 voice.gain.tx.analog.chassis=3 voice.gain.tx.analog.chassis.IP_300=0 voice.gain.tx.analog.chassis.IP_4000=3 voice.gain.tx.analog.chassis.IP_601=0 voice.gain.tx.digital.handset=0 voice.gain.tx.digital.headset=0 voice.gain.tx.digital.chassis=3 voice.gain.tx.digital.chassis.IP_4000=0 voice.gain.tx.digital.chassis.IP_601=6 voice.gain.tx.analog.preamp.handset=14 voice.gain.tx.analog.preamp.headset=23 voice.gain.tx.analog.preamp.chassis=32 voice.gain.tx.analog.preamp.chassis.IP_601=32/ -- E ___ --Bandwidth 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] new beta Grandstream firmware HT488_496_386
I'm almost afraid to ask, but is the HT 386 known for having a lot of troubles? I just installed one at home about 2 weeks ago, and knock on wood, it's only locked up once, and this was when I was still in the process of tweaking the config to work optimally w/ [EMAIL PROTECTED] I can't say I'm entirely pleased with the slight echo and buzz I'm detecting, but so far it's at least worked.. This isn't the consensus though, huh?! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: sábado, 4 de Março de 2006 0:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 They promised me this for my POS 386 adapters that need to be rebooted every few days from lockups about 4 months ago. Gee I wonder if this will work. Probably not. On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote: http://grandstream.com/BETATEST/HT488_496_386/ winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to change Budgetone dialtone?
Hi try http://www.grandstream.com/y-downloads.htm Download the IP Phone Custom Ringtones Generation Tool Unzip and read the readme Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dmitry Ivanov Sent: 07 March 2006 13:40 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to change Budgetone dialtone? Good day! Is is possible to change dialtone (and other tones as well) in BT-102? ### 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] Problem ChanSpy
Hi list, I got a question: When I try to ChanSpy a SIP channel I only listen one channel, for example, I call from 302 extension and I have two active channels: SIP/r1-voip-1b7b (None) Up Bridged Call(SIP/302-f1f1) SIP/302-f1f1 [EMAIL PROTECTED] Up Dial(SIP/[EMAIL PROTECTED]|4 When I try to spy this call from another extension: 1.SIP/301-fecc [EMAIL PROTECTED] Up ChanSpy(SIP/302) 2.SIP/r1-voip-1b7b (None) Up Bridged Call(SIP/302-f1f1) 3.SIP/302-f1f1 [EMAIL PROTECTED] Up Dial(SIP/[EMAIL PROTECTED]|4 I got 3 active channels, the one spying, the one that places the call and the one that receives the call. My problem is in the spying channel I can only hear the one that receives the call (3) but I cannot hear the channel (2): In the file sip.conf : [302] canreinvite=yes [301] canreinvite=yes Thanks for your help, ___ --Bandwidth 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 ChanSpy
I al surprised that you are hearing anything at all. the setting you have in your sip.conf ionstructs * to allow the end-points to send the 'voice' directly betrween them. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Guarnido Sent: Tuesday, March 07, 2006 8:57 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Problem ChanSpy Hi list, I got a question: When I try to ChanSpy a SIP channel I only listen one channel, for example, I call from 302 extension and I have two active channels: SIP/r1-voip-1b7b (None) Up Bridged Call(SIP/302-f1f1) SIP/302-f1f1 [EMAIL PROTECTED] Up Dial(SIP/[EMAIL PROTECTED]|4 When I try to spy this call from another extension: 1.SIP/301-fecc [EMAIL PROTECTED] Up ChanSpy(SIP/302) 2.SIP/r1-voip-1b7b (None) Up Bridged Call(SIP/302-f1f1) 3.SIP/302-f1f1 [EMAIL PROTECTED] Up Dial(SIP/[EMAIL PROTECTED]|4 I got 3 active channels, the one spying, the one that places the call and the one that receives the call. My problem is in the spying channel I can only hear the one that receives the call (3) but I cannot hear the channel (2): In the file sip.conf : [302] canreinvite=yes [301] canreinvite=yes Thanks for your help, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to change Budgetone dialtone?
On Tuesday 07 March 2006 15:49, Lee Archer wrote: Download the IP Phone Custom Ringtones Generation Tool Unzip and read the readme Ringtone != dialtone. pgpEye3ebfT7t.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to change Budgetone dialtone?
Sorry... Just ignore me. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dmitry Ivanov Sent: 07 March 2006 14:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to change Budgetone dialtone? On Tuesday 07 March 2006 15:49, Lee Archer wrote: Download the IP Phone Custom Ringtones Generation Tool Unzip and read the readme Ringtone != dialtone. ### 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] Advice on configuration
Hi All, I am looking to see if this is possible and any pointers if it is. It seems straight forward but not too sure. I have 4 extensions 2000 to 2003 I have one voip external account with Sipdiscount. I want any of the 4 extensions to share that single sipdiscount account. I also have 2 voip incoming numbers through another company (sipgate). I want one of these to ring 3 phones and the other one to ring the 4th extension if dialled. Is that possible? Thanks Paul ___ --Bandwidth 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] new beta Grandstream firmware HT488_496_386
It will lockup on you every 10 days or so and if you use both ports and have calls at the same time then good luck. Does your call waiting work on both ports? Mine does not. On 3/7/06, Steve Jones [EMAIL PROTECTED] wrote: I'm almost afraid to ask, but is the HT 386 known for having a lot of troubles? I just installed one at home about 2 weeks ago, and knock on wood, it's only locked up once, and this was when I was still in the process of tweaking the config to work optimally w/ [EMAIL PROTECTED] I can't say I'm entirely pleased with the slight echo and buzz I'm detecting, but so far it's at least worked.. This isn't the consensus though, huh?! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: sábado, 4 de Março de 2006 0:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 They promised me this for my POS 386 adapters that need to be rebooted every few days from lockups about 4 months ago. Gee I wonder if this will work. Probably not. On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote: http://grandstream.com/BETATEST/HT488_496_386/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth 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 problem with call duration
Regards!
During the use of areski a2billing software
I'm getting same problem all the time.
Actually, after 15 minutes ofspeaking
to someone over calling card, connection brakes.
Installation was as smooth as it could be
so I don't think I made same kind of a mess in that domain. This is the only
problem in the aplication.
In the logs everything seems to be
fine.
I'am sending You log as an apendix bellow
the text.
Is it a asterisk problem
or...
a2billing.log
[05/03/2006
19:39:45]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_rate-engine: Count Total
result 1][05/03/2006
19:39:45]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_rate-engine: Count Total
result 1][05/03/2006
19:39:45]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_rate-engine: number_trunk
1][05/03/2006
19:39:46]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT
(81.1667)][05/03/2006
19:39:46]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT: k=0
- res_calcultimeout:4869][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:DIAL
SIP/odlazni/00436642780018|90|HL(4869000:61000:3)[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[K=0]:[ANSWEREDTIME=751-DIALSTATUS=ANSWER][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[USEDRATECARD=0][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_CALCULCOST: K=0 -
CALLDURATION:751][05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[TEMP
- CC_RATE_ENGINE_CALCULCOST: 1. COST: -12.5167]:[ (751/60) * 1
][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_CALCULCOST: K=0 - FINAL
COST: -12.5167][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_UPDATESYSTEM:
usedratecard K=0 - (sessiontime=751 :: dialstatus=ANSWER ::
cost=12.5167)][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_stop 1.1: SQL: INSERT INTO
call (uniqueid,sessionid,username,nasipaddress,starttime,sessiontime,
calledstation, terminatecause, stoptime, calledrate, sessionbill,
calledcountry, calledsub, destination, id_tariffgroup, id_tariffplan,
id_ratecard, id_trunk, src) VALUES ('1141583953.47',
'SIP/callingcard-50e8', '0474', '', CURRENT_TIMESTAMP - INTERVAL 751
SECOND , '751', '00436642780018', 'ANSWER', now(), '1', '12.5167',
'', '', 'svet', '1', '1', '1', '1', '051359687' )][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_stop 1.1: SQL:
DONE][05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_stop
1.2: SQL: UPDATE cc_card SET credit= credit-12.5167,
redial='00436642780018', lastuse=now(), nbused=nbused+1 WHERE
username='0474'][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[callingcard_acct_stop][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CHANNEL STATUS : 6 = Line is
up][05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[CREDIT STATUS :
68.6500][05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[DTMF
DESTINATION :: -1][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CHANNEL STATUS : -1 = There is no
channel that matches SIP/callingcard-50e8][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CREDIT STATUS :
68.6500][05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[Start:
UPDATE cc_card SET inuse=inuse-1 WHERE username='0474'][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[STOP - EXIT][05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_rate-engine: Count Total
result 1][05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_rate-engine: Count Total
result 1][05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_rate-engine: number_trunk
1][05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT
(46.7500)][05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT: k=0
- res_calcultimeout:2804][05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:DIAL
SIP/odlazni/00497720954992|90|HL(2804000:61000:3)[05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[K=0]:[ANSWEREDTIME=937-DIALSTATUS=ANSWER] ( my max call duration
937 sec)[05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[USEDRATECARD=0][05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_CALCULCOST: K=0 -
CALLDURATION:937][05/03/2006 19:53:58]:[CallerID:051212072]:[CN:4513]:[TEMP
- CC_RATE_ENGINE_CALCULCOST: 1. COST: -15.6167]:[ (937/60) * 1
][05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_CALCULCOST: K=0 - FINAL
COST: -15.6167][05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_UPDATESYSTEM:
usedratecard K=0 - (sessiontime=937 :: dialstatus=ANSWER ::
cost=15.6167)][05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_stop 1.1: SQL: INSERT INTO
call (uniqueid,sessionid,username,nasipaddress,starttime,sessiontime,
calledstation, terminatecause, stoptime, calledrate, sessionbill,
[Asterisk-Users] Send One Touch Record to mail
How can I send recordings, that I have recorded with One Touch Record, to e-mail address that is defined in voicemail.conf? Thank you for your ideas. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth 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] nwebmail
Hi all, I got also your question, how to use nwebmail? Nwebmail is used for administration mail reports, i think. Take a look on this: http://www.vozdigital.org/modules.php?op=modloadname=Newsfile=articlesid=95 I've made login with user: admin password: mypassword_for_admin I'm developing a solution based on [EMAIL PROTECTED], and also trying to improve administration docs for my client, would be an extra value to understand what for nwebmail and main advantages... Basically it seems to be that the main cronjobs and main events are there on email messages, am I wrong? Best regards, Marco Mouta On 1/18/06, yrving rivas [EMAIL PROTECTED] wrote: Ok, thanks, it works for me. Regards, Yrving Dovid Bender [EMAIL PROTECTED] escribió: If you are new I would reccomend using [EMAIL PROTECTED] http://asteriskathome.soundforge.net . It is a great resource for beginers. Also get the book (again I dont have the URL if some one does please post it). Asterisk Regards, Dovid --- yrving rivas wrote: Hello! I am new to Asterisk, AMP, Linux...did I say all?.. I just installed Asterisk, and for my needs it is working great. In my AMP I see the nwebmail but I can´t get into it. When I place my login and password, comes with the following message: An internal error has occured. Please co ntact your system administrator. If you are the system administrator, check the log files. The log files don´t help me very much. Can someone tell me how to use the nwebmail?, how to get in for first time? Regards! Yrving - Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 Do You Yahoo!? La mejor conexión a Internet y 2GB extra a tu correo por $100 al mes. http://net.yahoo.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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] and H323
On Tue, 2006-03-07 at 12:08 +0200, Viktor Tatianin wrote: Hello I attempt installing H323 at my [EMAIL PROTECTED] for this use asteriskathome-h323-1.0.zip but have next problem chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_show_channels': If you have asterisk 1.2.4 version you must have to compile oh323 as in http://www.oinko.net/astrecipes/index.php?n=40 but replacing the versions from: http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/pwlib-Mimas_patch2-src-tar.gz http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/openh323-Mimas_patch2-src-tar.gz http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.7.3.tar.gz Please help for resolve this problem Viktor Tatianin ___ --Bandwidth 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. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --Bandwidth 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] Destroying a SIP extension doesn't destroy voicemail box?is this a bug?
Hi all, I'm using [EMAIL PROTECTED] 2.5, and i've done: 1-Create a SIP extension. 2-Leave there a Voicemail message 3-Remove SIP extension Then I've create another SIP extension but with the same number of the above one. I found imediately a voicemail message in my voicemail box. Is this a bug? Am I doing something wrong? Best regards, Marco Mouta ___ --Bandwidth 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] PBX-VPN-SIP-Asterisk trouble
Hi all! I have the following setup: Phone lines - traditional PBX - Welltech 3802 - VPN - Asterisk - Linksys PAP2/Welltech ATA-151 - phone There is 2 pieces of Welltech 3802 (2 port FXO) connected to 4 (2x2) PBX extensions. Asterisk is a proxy here. Each device successfully register itself. I tried the setup above with Linksys and Welltech devices as well. I setup Asterisk as a local PBX and phones can call each others on Asterisk side and possible transfering calls. I setup Welltech 3802 with hotline mode so if someone call the public number from outside the call transferred through VPN and phone rings in front of me. Great. It's still possible transfer call within Asterisk side. Excellent. The problem comes when I want to call extension on PBX side or transfer incoming call to the PBX side. I got the line sound when I press flash, the caller hear the MOH and when I call extensions on PBX side I got only busy tone. How could I tell that Asterisk send back the flashDTMF on the same PBX extension where call comes from? I think this is important for PBX to connect lines inside right. How could I route outcoming calls on a port of Welltech 3802? An example (because my grammar is hard to understand :-) Call from outside 1, PBX rings on connected Welltech 3802 port 2, Welltech 3802 picks up the phone and transfer to the specified hotline number 3, (packets going through the VPN) 4, Linksys got an INVITE from Welltech and starts ringing phone 5, I pickup the phone and talk A (local extension): 6, I press flash and got a line tone 7, I enter the digits of local extension (4 digits) 8, Asterisk search in registered peers and found it 9, Asterisk connect to SIP device phone rings there B (external extension): 6, I press flash and got a line tone 7, I enter the digits of local extension (3 digits) 8, Asterisk send a flash and the entered digits back to PBX via the Welltech 3802 9, PBX connect to the specified extension phone rings there My question is how could I tell to Asterisk send back flash and DTMF to PBX on the active connection? I'm stand at the B/8 and don't know what I do. I digging the internet in last 2 days but no solution. bye, Zsolt ___ --Bandwidth 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] call manager integration
On Mon, 2006-03-06 at 15:42, Jerry Geis wrote: / here is some of the output. I am no longer the to spcifically do sip // debug but this is what I have. // along with my sip.conf snip. // // The call to extension 3726 never rings. so it never gets answered. // / Are you sure your sip trunk and route pattern are in the same partition/CSS by chance? Without more info (AGI script and SIP debug), I really can't be much more help. Your sip.conf entry is good though. Your callmanager context from extensions.conf will help as well. -Greg Greg, here is the sip debug output... Again I can call into the asterisk box but cant call out with call files. You mentioned my sip.conf entry looked OK and I have canreinvite=yes in that file for the CallManager. Thanks, Jerry sip debug SIP Debugging re-enabled -- Attempting call on SIP/CallManager//3726 for [EMAIL PROTECTED]:1 (Retry 1) We're at 10.101.69.200 port 12592 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 10.101.66.10:5060: INVITE sip:/[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK32e866aa;rport From: Admin System 34 sip:[EMAIL PROTECTED];tag=as2d52e2ca To: sip:/[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 07 Mar 2006 14:49:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 4082 4082 IN IP4 10.101.69.200 s=session c=IN IP4 10.101.69.200 t=0 0 m=audio 12592 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- co-drpage-01*CLI -- SIP read from 10.101.66.10:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK32e866aa;rport From: Admin System 34 sip:[EMAIL PROTECTED];tag=as2d52e2ca To: sip:/[EMAIL PROTECTED];tag=33558825 Date: Tue, 07 Mar 2006 14:49:34 GMT Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 --- (9 headers 0 lines)--- co-drpage-01*CLI -- SIP read from 10.101.66.10:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK32e866aa;rport From: Admin System 34 sip:[EMAIL PROTECTED];tag=as2d52e2ca To: sip:/[EMAIL PROTECTED];tag=33558825 Date: Tue, 07 Mar 2006 14:49:34 GMT Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 --- (9 headers 0 lines)--- Transmitting (no NAT) to 10.101.66.10:5060: ACK sip:/[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK32e866aa;rport From: Admin System 34 sip:[EMAIL PROTECTED];tag=as2d52e2ca To: sip:/[EMAIL PROTECTED];tag=33558825 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Channel SIP/CallManager-a48d was never answered. Mar 7 08:49:32 WARNING[5219]: cdr.c:548 ast_cdr_disposition: Cause not handled -- Executing AGI(OutgoingSpoolFailed, smvoice|-digium_failed) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice Destroying call '[EMAIL PROTECTED]' == Spawn extension (smvoice-dialout, failed, 1) exited non-zero on 'OutgoingSpoolFailed' Mar 7 08:49:34 NOTICE[5219]: pbx_spool.c:270 attempt_thread: Call failed to go through, reason 8 -- Attempting call on SIP/CallManager//3726 for [EMAIL PROTECTED]:1 (Retry 1) We're at 10.101.69.200 port 19812 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 10.101.66.10:5060: INVITE sip:/[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK00e00a23;rport From: Admin System 34 sip:[EMAIL PROTECTED];tag=as4fdb4cfa To: sip:/[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 07 Mar 2006 14:49:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 4082 4082 IN IP4 10.101.69.200 s=session c=IN IP4 10.101.69.200 t=0 0 m=audio 19812 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- co-drpage-01*CLI -- SIP read from 10.101.66.10:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.101.69.200:5060;branch=z9hG4bK00e00a23;rport From: Admin System 34 sip:[EMAIL PROTECTED];tag=as4fdb4cfa To: sip:/[EMAIL PROTECTED];tag=33558827 Date: Tue, 07 Mar 2006 14:49:46 GMT Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 --- (9 headers 0 lines)---
[Asterisk-Users] Asterisk + SE Linux
Hi guys, I am busy planning to implement SE Linux on my asterisk box. Either that or I will use AppArmor from Suse. I just want to know what are others experiences/incidents with SE Linux or AppArmor thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk download file locations
Or hardcode the Digium URL in your script and on failure grab from your mirrors, and to make absolutely sure your mirror should resolve to a DNS name and then if *that* fails, a hardcoded IP. That way, you get 3 layers. -Original Message- From: Joseph Tanner [mailto:[EMAIL PROTECTED] Sent: Monday, March 06, 2006 9:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk download file locations If it's a commercial product, you should definitely mirror the files. Not only because you're benefiting financially, but because you need full control. Perhaps you'd like to incorporate a patch or two in the source? Or maybe you'd like to use a stable label, so the script downloads stable.tar.gz. Once you've tested a new version and it works with your customizations/patches/whatever, you just upload it and rename it as stable.tar.gz, and any customer who runs your script automatically gets the latest and greatest. You could simulate some of this without mirroring asterisk though. Have the script check your server for a value, say the location to download asterisk. This will let you update the URL if it changes, or have it point to a newer version of asterisk, etc. Of course, I would hardcode in some values that the script could use, in case it can't reach your server but can reach digium's. Just some thoughts. Joseph Tanner On 3/6/06, Peter Fern [EMAIL PROTECTED] wrote: Still, if you mirror them yourself, this problem all but goes away. Alistair Cunningham wrote: Colin, Because having the logic is not the correct thing to do from an engineering point of view. Consider: - What if Digium change the directory structure again? Having a published directory structure is the elegant thing to do. - Not only does it break build scripts but it breaks search engines too. - Our scripts already have more conditional logic than I'm happy with, dealing with all the inconsistencies that Linux distributions throw at us. Anything which makes the installation process less brittle is a good thing. Alistair Cunningham, Integrics Ltd, +44 20 799 39 799 sip:[EMAIL PROTECTED] http://integrics.com/ Colin Anderson wrote: Why wouldn't you build in trivial conditional logic into your script or mirror the Asterisk builds yourself? -Original Message- From: Alistair Cunningham [mailto:[EMAIL PROTECTED] Sent: Monday, March 06, 2006 8:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk download file locations This is a request to the website manager for asterisk.org. The build scripts for our ITSP product include the URLs to download the Asterisk files, such as: wget http://ftp.digium.com/pub/asterisk/asterisk-1.2.5.tar.gz; However, if a new version is released, asterisk-1.2.5.tar.gz is moved to the old directory. This breaks our scripts until we can update them and send them to our resellers. Would it be possible to have a fixed address for a particular asterisk release that will never (or at least not for a long time) change? Perhaps put all (except very old) versions in the same directory, with a 'latest' link to the latest one? ___ --Bandwidth 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 ___ --Bandwidth 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] Destroying a SIP extension doesn't destroyvoicemail box?is this a bug?
Whey you 'destroy' a Sip extension you are only removing the entrys that allow you to make and receive the auth needed to do so. Your voicemail files are not tied to an extension but are independent and are only 'married' when you specify it in your sip.conf or other channel configs. Removing a confi from a channel will not touch voicemail. You need to go into the voicemail.conf and/or the voicemail spool directories and remove the entries or files yourself. AAH should do this as part of the script. But it does not, would probably cause more harm than good. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Marco Mouta Sent: Tuesday, March 07, 2006 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Destroying a SIP extension doesn't destroyvoicemail box?is this a bug? Hi all, I'm using [EMAIL PROTECTED] 2.5, and i've done: 1-Create a SIP extension. 2-Leave there a Voicemail message 3-Remove SIP extension Then I've create another SIP extension but with the same number of the above one. I found imediately a voicemail message in my voicemail box. Is this a bug? Am I doing something wrong? Best regards, Marco Mouta ___ --Bandwidth 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] Toll free nos
Hello everyone, I am in need of 20 US toll free nos and 10 non toll free nos, termination using IAX. Are there any reliable companies that you can recommend? Thank you With regards, San ___ --Bandwidth 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] webvmail
My question is about webvmail, not nwebvmail. I have never used AMP (seems like cheating). My question is in regards to plain jane Asterisk install. Just like making samples after you compile asterisk you are able to make webvmail. Basically it is a interface into the voicemail system fro the web. I have apache installed on Fedora and am able to bring up the localhost test page. When I try to open vmail.cgi from the browser nothing happens. As I stated earlier I dont know whether this is even what I am looking for. I believe the app compiled correctly as I got no errors. Can anyone point me in the right direction? Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 ___ --Bandwidth 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] Advice on configuration
Hi Paul I am looking to see if this is possible and any pointers if it is. It seems straight forward but not too sure. I have 4 extensions 2000 to 2003 I have one voip external account with Sipdiscount. I want any of the 4 extensions to share that single sipdiscount account. 'share' as in dial out through? Assuming they're SIP phones several ways to do it, here's my favourite sip.conf [phone1] .. context=sipphones ... [phone2] .. context=sipphones ... [sipdiscount] stuff about your sipdiscount account extensions.conf [sipphones] other-things-you-want-them-to-be-able-to-dial include = sipdiscount-outbound [sipdiscount-outbound] exten = somepattern,1,Dial([EMAIL PROTECTED]) etc I also have 2 voip incoming numbers through another company (sipgate). I want one of these to ring 3 phones and the other one to ring the 4th extension if dialled. Is that possible? Yep sip.conf register =:[EMAIL PROTECTED]/111 register =mmm:[EMAIL PROTECTED]/222 [sipgate] type=friend host=sipgate.co.uk insecure=very context=sipgate-inbound extensions.conf [sipgate-inbound] exten = 11,1,Dial(SIP/2000SIP/2001SIP/2002) exten = 22,1,Dial(SIP/2003) Give me a shout if you want more help Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth 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] indications SIP
Apologies if this is an old question; I've searched the list and the wiki but have not been able to find a definitive answer. I have an Aastra 480i phone registered with * 1.2.4; I want to generate UK ringback tones when the handset dials another internal extension. On my Zap channels, I have this in place by editing /etc/zaptel.conf; however I've had no luck with the Sip handset (I have the same problem with a Grandstream ATA). My indications.conf has country=uk and I've also set both the general and extension sections in sip.conf to language=uk, but I still only get US ringback tones on the Aastra handset. I've probably missed some vital point, but I'd appreciate any pointers people could give. Regards, Chris -- Chris Notley [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] Oh this is bad.... bindaddr and rtp traffic
I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp 0 788 0.0.0.0:5060 0.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A 'rtp debug' shows it's receiving the RTP packets, it just seems it does nothing with them. Anyone seen this? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic
On Tue, 7 Mar 2006 09:12:25 -0700 Douglas Garstang [EMAIL PROTECTED] wrote: I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp0788 0.0.0.0:50600.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A 'rtp debug' shows it's receiving the RTP packets, it just seems it does nothing with them. Anyone seen this? Doug. I thought all RTP was controlled through rtp.conf and only the SIP traffic was controlled through SIP.conf. I am not sure what settings, beside the RTP port range, you can out into the rtp.conf though. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PLEASE HELP ,a2billing problem with call duration
Regards!
During the use of areski a2billing software
I'm getting same problem all the time.
Actually, after 15 minutes ofspeaking
to someone over calling card, connection brakes.
Installation was as smooth as it could be
so I don't think I made same kind of a mess in that domain. This is the only
problem in the aplication.
In the logs everything seems to be
fine.
I'am sending You log as an apendix bellow
the text.
Is it a asterisk problem
or...
a2billing.log
[05/03/2006
19:39:45]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_rate-engine: Count Total
result 1][05/03/2006
19:39:45]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_rate-engine: Count Total
result 1][05/03/2006
19:39:45]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_rate-engine: number_trunk
1][05/03/2006
19:39:46]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT
(81.1667)][05/03/2006
19:39:46]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT: k=0
- res_calcultimeout:4869][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:DIAL
SIP/odlazni/00436642780018|90|HL(4869000:61000:3)[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[K=0]:[ANSWEREDTIME=751-DIALSTATUS=ANSWER][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[USEDRATECARD=0][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_CALCULCOST: K=0 -
CALLDURATION:751][05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[TEMP
- CC_RATE_ENGINE_CALCULCOST: 1. COST: -12.5167]:[ (751/60) * 1
][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_CALCULCOST: K=0 - FINAL
COST: -12.5167][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_UPDATESYSTEM:
usedratecard K=0 - (sessiontime=751 :: dialstatus=ANSWER ::
cost=12.5167)][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_stop 1.1: SQL: INSERT INTO
call (uniqueid,sessionid,username,nasipaddress,starttime,sessiontime,
calledstation, terminatecause, stoptime, calledrate, sessionbill,
calledcountry, calledsub, destination, id_tariffgroup, id_tariffplan,
id_ratecard, id_trunk, src) VALUES ('1141583953.47',
'SIP/callingcard-50e8', '0474', '', CURRENT_TIMESTAMP - INTERVAL 751
SECOND , '751', '0043***', 'ANSWER', now(), '1',
'12.5167', '', '', 'svet', '1', '1', '1', '1', '051359687'
)][05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_stop
1.1: SQL: DONE][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_stop 1.2: SQL: UPDATE
cc_card SET credit= credit-12.5167, redial='00436642780018',
lastuse=now(), nbused=nbused+1 WHERE username='0474'][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[callingcard_acct_stop][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CHANNEL STATUS : 6 = Line is
up][05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[CREDIT STATUS :
68.6500][05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[DTMF
DESTINATION :: -1][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CHANNEL STATUS : -1 = There is no
channel that matches SIP/callingcard-50e8][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CREDIT STATUS :
68.6500][05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[Start:
UPDATE cc_card SET inuse=inuse-1 WHERE username='0474'][05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[STOP - EXIT][05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_rate-engine: Count Total
result 1][05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_rate-engine: Count Total
result 1][05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_rate-engine: number_trunk
1][05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT
(46.7500)][05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT: k=0
- res_calcultimeout:2804][05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:DIAL
SIP/odlazni/00497720954992|90|HL(2804000:61000:3)[05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[K=0]:[ANSWEREDTIME=937-DIALSTATUS=ANSWER] ( my max call duration
937 sec)[05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[USEDRATECARD=0][05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_CALCULCOST: K=0 -
CALLDURATION:937][05/03/2006 19:53:58]:[CallerID:051212072]:[CN:4513]:[TEMP
- CC_RATE_ENGINE_CALCULCOST: 1. COST: -15.6167]:[ (937/60) * 1
][05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_CALCULCOST: K=0 - FINAL
COST: -15.6167][05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_UPDATESYSTEM:
usedratecard K=0 - (sessiontime=937 :: dialstatus=ANSWER ::
cost=15.6167)][05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_stop 1.1: SQL: INSERT INTO
call (uniqueid,sessionid,username,nasipaddress,starttime,sessiontime,
calledstation, terminatecause, stoptime, calledrate, sessionbill,
[Asterisk-Users] I can't receive multiple pages with spandsp
Hi all,
I'trying to use spandsp (app_rxfax) to receive faxes.
When there are more than one page, the system creates a tiff file with only the
first page and the other are lost, even if the full log says:
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c:
==
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Pages transferred: 1
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Image size: 1728 x 1118
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Image resolution7700 x 3850
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Transfer Rate: 9600
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Bad rows0
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Longest bad row run 0
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Compression type1
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c: Image size (bytes) 0
Mar 7 17:17:42 DEBUG[5876] app_rxfax.c:
==
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c:
==
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Pages transferred: 2
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Image size: 1728 x 1117
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Image resolution7700 x 3850
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Transfer Rate: 9600
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Bad rows0
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Longest bad row run 0
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Compression type1
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c: Image size (bytes) 0
Mar 7 17:18:13 DEBUG[5876] app_rxfax.c:
==
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c:
==
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Fax successfully received.
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Remote station id: 3002
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Local station id:
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Pages transferred: 2
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Image resolution: 7700 x 3850
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c: Transfer Rate: 9600
Mar 7 17:18:16 DEBUG[5876] app_rxfax.c:
==
In extensions.conf I have:
exten = 1080,1,NoOp(Entro nel context from-FAX)
exten = 1080,2,Answer
exten = 1080,3,Macro(ricezionefax)
exten = 1080,4,system(tiff2ps -2 -a -e -z -w 8 -h 10.5 ${FAXFILE} | lpr [EMAIL
PROTECTED]) ;;;I send the fax to my printer
exten = 1080,5,Hangup
and
[macro-ricezionefax]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,rxfax(${FAXFILE})
exten = s,102,Goto(2)
Is this a problem of spandsp (I'm using spandsp-0.0.2pre25)
or is there an error in my configuration?
Thanks in advance,
Marco.
___
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RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic
Asterisk does not like multiple interfaces in the way you are configured. You can either: A) use the bindaddr in the sip.conf to limit where the packsge come and go. B) use an outside traffic manager Look up the archives, kpf explained why this would not work, as asterisk can't do load balancing at this time -Original Message- From: Robert Webb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 3/7/06 11:27 AM Subject: Re: [Asterisk-Users] Oh this is bad bindaddr and rtp traffic On Tue, 7 Mar 2006 09:12:25 -0700 Douglas Garstang [EMAIL PROTECTED] wrote: I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp0788 0.0.0.0:50600.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A 'rtp debug' shows it's receiving the RTP packets, it just seems it does nothing with them. Anyone seen this? Doug. I thought all RTP was controlled through rtp.conf and only the SIP traffic was controlled through SIP.conf. I am not sure what settings, beside the RTP port range, you can out into the rtp.conf though. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Oh this is bad.... bindaddr and rtp traffic
Pardon my candour, but for a product Digium calls 'enterprise grade' it sure seems to be missing a few features. -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 07, 2006 9:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Oh this is bad bindaddr and rtp traffic Asterisk does not like multiple interfaces in the way you are configured. You can either: A) use the bindaddr in the sip.conf to limit where the packsge come and go. B) use an outside traffic manager Look up the archives, kpf explained why this would not work, as asterisk can't do load balancing at this time -Original Message- From: Robert Webb [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 3/7/06 11:27 AM Subject: Re: [Asterisk-Users] Oh this is bad bindaddr and rtp traffic On Tue, 7 Mar 2006 09:12:25 -0700 Douglas Garstang [EMAIL PROTECTED] wrote: I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp0788 0.0.0.0:50600.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A 'rtp debug' shows it's receiving the RTP packets, it just seems it does nothing with them. Anyone seen this? Doug. I thought all RTP was controlled through rtp.conf and only the SIP traffic was controlled through SIP.conf. I am not sure what settings, beside the RTP port range, you can out into the rtp.conf though. Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] res_mysql.conf DNS SRV lookup
Hi friends, I am using Real Time Asterisk Architecture where I have put the Sip users/peers and extensions defining the dialplan in tables in a mysql database. Currently, asterisk points to my single database server as configured: -- /etc/asterisk/res_mysql.conf -- [general] dbhost = serdb1.goldline.net dbname = asterisk dbuser = asterisk dbpass = asterisk-arcph0n3 dbport = 3306 dbsock = /tmp/mysql.sock But what I want to do is to set dbhost in /etc/asterisk/res_mysql.conf to point asterisk to a DNS SRV record so that I can implement mysql redundancy. I defined the SRV record in our DNS server and put it in dbhost field in /etc/asterisk/res_mysql.conf but asterisk wouldnt start up! Can anyone tell me if asterisk mysql drivers support DNS SRV records lookup?! If not, how can I achieve this?! Thanks ramin ___ --Bandwidth 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] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3 WARNING!!!
Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated bootrom, nice. Rebooted again. Updated sip firmware. Also nice. Upon the next reboot, the wheels started falling off. The phones would not get changes I made to any of the .cfg files. Several phones would take 20 minutes or more to boot, only to display a 0x4000 config file error. What happened? I have been using various Polycom's with AstLinux (and vsftpd 2.0.3 that I include with it) for quite some time, with no problems whatsoever. Until now. I had been running bootrom 3.0.1 and various versions of the SIP image at several other sites with no problem. At this point I was still unable to accept the fact that I might not be able to run this latest bootrom. After many trial and tribulations, I finally rsync'ed (with -avr) the FTP directory from the AstLinux machine to my laptop running CentOS 4. I configured the vsftpd daemon (version 2.0.1) IDENTICALLY (with the exception of PAM and TCP wrappers) and crossed my fingers... After re-configuring the IP 501 to use my laptop, imagine my surprise when the most problematic of them booted right away without problems. Again and again, everything was fine. So now I just had to break out ethereal and see what was going on. While I have not completely finished my analysis, it appears that Polycom firmware 3.1.3 bombs out when transferring files with vsftpd 2.0.3. The symptom appears to be repeated TCP SYNs from the Polycom to the ftp daemon on port 20. The Polycom will keep retrying and increment its source port number by one every few minutes. Like I said, I need to dig into this more, but I figured I'd report what I know and see if anyone out there can fill in the holes. Here's what I did. It appears that BootRom 3.1.3 works with vsftpd 2.0.1, so I placed bootrom 3.0.1 (which I know works with vsftpd 2.0.3) on my CentOS server and downgraded the phone to 3.0.1. I then placed 3.0.1 and SIP app 1.6.5 (which I was using the whole time, btw) on my AstLinux server running vsftpd 2.0.3. All was good. So now I am successfully running with the following: Polycom IP 501 Bootrom 3.0.1 SIP 1.6.5 AstLinux 0.3.7 vsftpd 2.0.3 I will also try to fix (or workaround) this by trying the following: upgrading AstLinux to include vsftpd 2.0.4 trying an intermediate BootRom release between 3.0.1 and 3.1.3 (find out exactly where/when it broke) trying an even newer Polycom BootRom when it becomes available upgrading the kernel in AstLinux (I doubt that's it) fiddling with iptables rules in AstLinux (iptables was loaded, but obviously 3.0.1 doesn't have a problem with it) This also might be related to the problems described here: http://forums.digium.com/viewtopic.php?p=14847sid=6e70577c37bd345cfc164a01e64e113a Any thoughts? Comments? Suggestions? P.S. - I will be updating the Polycom config files at http://www.krisk.org/asterisk/pcom/ to reflect some new changes in this firmware release. I just need to get my phones working first :)! -- Kristian Kielhofner ___ --Bandwidth 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] webvmail
I compiled the newest version of * from cvs about a week ago on Fedora. I think all I had to do after issuing the make webvmail was install the perl and perl-suidperl packages. I got that information (and anything else I might have done but forgotten) by searching for webvmail.cgi at voip-info.org. On 3/7/06, Jordan Novak [EMAIL PROTECTED] wrote: My question is about webvmail, not nwebvmail. I have never used AMP (seems like cheating). My question is in regards to plain jane Asterisk install. Just like making samples after you compile asterisk you are able to make webvmail. Basically it is a interface into the voicemail system fro the web. I have apache installed on Fedora and am able to bring up the localhost test page. When I try to open vmail.cgi from the browser nothing happens. As I stated earlier I don't know whether this is even what I am looking for. I believe the app compiled correctly as I got no errors. Can anyone point me in the right direction? Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- kris seraphine ___ --Bandwidth 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] HW Echo Cancellers
Hi, In December, I posted an enquiry asking whether anyone had experience with the Sangoma A104d cards (see below) - I got a couple of responses, but basically it was that people have started playing with them, and would publish feedback at a later date. Does anyone have any further feedback at this stage? Many thanks in anticipation. Regards, Steve On 12/20/05, Steve Davies [EMAIL PROTECTED] wrote: http://www.google.com/search?q=cache:3AXi4YvnS80J:www.sangoma.com/company/news_releases/octasic.htmhl=en it seems that there will soon be an A102d, A104d and A108d available on the market. Given that only the A104d is available at present, can anyone give feedback on this product from an asterisk/end-user point of view? Is the EC any good? Does it solve your problems? Are the drivers stable? How is the voice quality? What size of server CPU did you use/need? etc etc etc... Any opinion would be useful to save us investing $$$s at this stage :). ___ --Bandwidth 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 compiling ztdummy on centos 4, 2.6 kernel
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Tuesday, March 07, 2006 6:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel On Tue, 2006-03-07 at 05:04 -0500, Sina Bahram wrote: However, as I pointed out in my email, that doesn't make any difference. If I leave it commented ... I get the exact same thing: just minus the make file error Same behavior, same error messages with the /etc scripts, the modprobe's and with everything else. Take care, Sina You could try the rpms at http://www.laimbock.com/asterisk/ Regards, Patrick ps please don't top post (put your answer *below* the posting). I could do that, yes. But I wanted to compile this software, and I am having trouble figuring out why something which is marked as stable is giving me such issues, after explicitly following all directions on a relatively standard setup. I will most definitely investigate the rpm's, but I would really like to compile this, so that I am not dependant on someone else for newer versions and so forth. Take care, Sina -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Tuesday, March 07, 2006 4:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Problem compiling ztdummy on centos 4, 2.6 kernel On Mon, 2006-03-06 at 10:54 -0500, Sina Bahram wrote: Here is the compilation process of zaptel I did edit the makefile and uncommented the #ztdummy, although, after I did that, I get the make error of ztdummy being defined more than once. [snip] You don't need to uncomment ztdummy in the Makefile because if you are using a 2.6 kernel it will be built automagically. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] res_mysql.conf DNS SRV lookup
Are you kidding? Asterisk doesn't do SRV. If you read all the VOIP books out there, SRV lookups are _the_ way to achieve redundancy. Digium hasn't gotten to it I guess with their 'enterprise class' product though. -Original Message-From: Ramin Nikaeen [mailto:[EMAIL PROTECTED]Sent: Tuesday, March 07, 2006 9:54 AMTo: asteriskUsersSubject: [Asterisk-Users] res_mysql.conf DNS SRV lookup Hi friends, I am using Real Time Asterisk Architecture where I have put the Sip users/peers and extensions defining the dialplan in tables in a mysql database. Currently, asterisk points to my single database server as configured: -- /etc/asterisk/res_mysql.conf -- [general] dbhost = serdb1.goldline.net dbname = asterisk dbuser = asterisk dbpass = asterisk-arcph0n3 dbport = 3306 dbsock = /tmp/mysql.sock But what I want to do is to set dbhost in /etc/asterisk/res_mysql.conf to point asterisk to a DNS SRV record so that I can implement mysql redundancy. I defined the SRV record in our DNS server and put it in dbhost field in /etc/asterisk/res_mysql.conf but asterisk wouldnt start up! Can anyone tell me if asterisk mysql drivers support DNS SRV records lookup?! If not, how can I achieve this?! Thanks ramin ___ --Bandwidth 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] HW Echo Cancellers
Steve, I just complete a setup of asterisk server in a production environment with a single A104D and there is no echo and the quality is okay. goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Tuesday, March 07, 2006 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HW Echo Cancellers Hi, In December, I posted an enquiry asking whether anyone had experience with the Sangoma A104d cards (see below) - I got a couple of responses, but basically it was that people have started playing with them, and would publish feedback at a later date. Does anyone have any further feedback at this stage? Many thanks in anticipation. Regards, Steve On 12/20/05, Steve Davies [EMAIL PROTECTED] wrote: http://www.google.com/search?q=cache:3AXi4YvnS80J:www.sangoma.com/company/ne ws_releases/octasic.htmhl=en it seems that there will soon be an A102d, A104d and A108d available on the market. Given that only the A104d is available at present, can anyone give feedback on this product from an asterisk/end-user point of view? Is the EC any good? Does it solve your problems? Are the drivers stable? How is the voice quality? What size of server CPU did you use/need? etc etc etc... Any opinion would be useful to save us investing $$$s at this stage :). ___ --Bandwidth 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] PLEASE HELP , a2billing problem with call duration
Hi d_pejic,
First of all, please never send several times the same question to the
list, it's really
not respectful for the others. Your issues should not pass in priority
from others.
As Kpfleming pointed out, Add-Ons/A2Billing are off topic for this
list, so please redirect
add-ons question to their authors. For A2billing I will set a forum/wiki in the
next days (with the new release).
About your issue, you can enable a2billing debug mode and send me the output
debug/what you see on the asterisk CLI.
KR, Areski
On 3/7/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Regards!
During the use of areski a2billing software I'm getting same problem all the
time.
Actually, after 15 minutes of speaking to someone over calling card,
connection brakes.
Installation was as smooth as it could be so I don't think I made same kind
of a mess in that domain. This is the only problem in the aplication.
In the logs everything seems to be fine.
I'am sending You log as an apendix bellow the text.
Is it a asterisk problem or...
a2billing.log
[05/03/2006
19:39:45]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_rate-engine:
Count Total result 1]
[05/03/2006
19:39:45]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_rate-engine:
Count Total result 1]
[05/03/2006
19:39:45]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_rate-engine:
number_trunk 1]
[05/03/2006
19:39:46]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT
(81.1667)]
[05/03/2006
19:39:46]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT:
k=0 - res_calcultimeout:4869]
[05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:DIAL
SIP/odlazni/00436642780018|90|HL(4869000:61000:3)
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[K=0]:[ANSWEREDTIME=751-DIALSTATUS=ANSWER]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[USEDRATECARD=0]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_CALCULCOST:
K=0 - CALLDURATION:751]
[05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[TEMP
- CC_RATE_ENGINE_CALCULCOST: 1. COST: -12.5167]:[ (751/60) * 1 ]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_CALCULCOST:
K=0 - FINAL COST: -12.5167]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_RATE_ENGINE_UPDATESYSTEM:
usedratecard K=0 - (sessiontime=751 :: dialstatus=ANSWER ::
cost=12.5167)]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_stop
1.1: SQL: INSERT INTO call
(uniqueid,sessionid,username,nasipaddress,starttime,sessiontime,
calledstation, terminatecause, stoptime, calledrate, sessionbill,
calledcountry, calledsub, destination, id_tariffgroup, id_tariffplan,
id_ratecard, id_trunk, src) VALUES ('1141583953.47', 'SIP/callingcard-50e8',
'0474', '', CURRENT_TIMESTAMP - INTERVAL 751 SECOND , '751',
'0043***', 'ANSWER', now(), '1', '12.5167', '', '',
'svet', '1', '1', '1', '1', '051359687' )]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_stop
1.1: SQL: DONE]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CC_asterisk_stop
1.2: SQL: UPDATE cc_card SET credit= credit-12.5167,
redial='00436642780018', lastuse=now(), nbused=nbused+1 WHERE
username='0474']
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[callingcard_acct_stop]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CHANNEL STATUS :
6 = Line is up]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CREDIT STATUS :
68.6500]
[05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[DTMF
DESTINATION :: -1]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CHANNEL STATUS :
-1 = There is no channel that matches SIP/callingcard-50e8]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[CREDIT STATUS :
68.6500]
[05/03/2006
19:52:37]:[CallerID:051359687]:[CN:0474]:[Start: UPDATE
cc_card SET inuse=inuse-1 WHERE username='0474']
[05/03/2006 19:52:37]:[CallerID:051359687]:[CN:0474]:[STOP
- EXIT]
[05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_rate-engine:
Count Total result 1]
[05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_rate-engine:
Count Total result 1]
[05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_asterisk_rate-engine:
number_trunk 1]
[05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT
(46.7500)]
[05/03/2006
19:38:06]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_ALL_CALCULTIMEOUT:
k=0 - res_calcultimeout:2804]
[05/03/2006 19:53:58]:[CallerID:051212072]:[CN:4513]:DIAL
SIP/odlazni/00497720954992|90|HL(2804000:61000:3)
[05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[K=0]:[ANSWEREDTIME=937-DIALSTATUS=ANSWER]
( my max call duration 937 sec)
[05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[USEDRATECARD=0]
[05/03/2006
19:53:58]:[CallerID:051212072]:[CN:4513]:[CC_RATE_ENGINE_CALCULCOST:
K=0 -
Re: [Asterisk-Users] most common VOIP echo simulaton for research purposes ?
thats good to hear .but there are so many digium cards that does echo distortion then why do you want to do this Giridhar Bandi On 3/7/06, Robert Rozman [EMAIL PROTECTED] wrote: Hi,I'm speech recognition researcher and would like to do some research onrecognition robustness in echo distortion of speech signal. Since VOIP isbecoming wide spread, I'd like to simulate (one or more) common echo distortions that mostly appear in voip communications ? Any example, FIR orIIR filter or acoustical system response ?Any other distortion worth researching ?Thanks in advance,regards, Rob.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3 WARNING!!!
I spent a weekend battling similar issues with 501s, using FC4/ proftpd. I finally switched from FTP to HTTP. On Mar 7, 2006, at 9:53 AM, Kristian Kielhofner wrote: Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated bootrom, nice. Rebooted again. Updated sip firmware. Also nice. Upon the next reboot, the wheels started falling off. The phones would not get changes I made to any of the .cfg files. Several phones would take 20 minutes or more to boot, only to display a 0x4000 config file error. What happened? I have been using various Polycom's with AstLinux (and vsftpd 2.0.3 that I include with it) for quite some time, with no problems whatsoever. Until now. I had been running bootrom 3.0.1 and various versions of the SIP image at several other sites with no problem. At this point I was still unable to accept the fact that I might not be able to run this latest bootrom. After many trial and tribulations, I finally rsync'ed (with -avr) the FTP directory from the AstLinux machine to my laptop running CentOS 4. I configured the vsftpd daemon (version 2.0.1) IDENTICALLY (with the exception of PAM and TCP wrappers) and crossed my fingers... After re-configuring the IP 501 to use my laptop, imagine my surprise when the most problematic of them booted right away without problems. Again and again, everything was fine. So now I just had to break out ethereal and see what was going on. While I have not completely finished my analysis, it appears that Polycom firmware 3.1.3 bombs out when transferring files with vsftpd 2.0.3. The symptom appears to be repeated TCP SYNs from the Polycom to the ftp daemon on port 20. The Polycom will keep retrying and increment its source port number by one every few minutes. Like I said, I need to dig into this more, but I figured I'd report what I know and see if anyone out there can fill in the holes. Here's what I did. It appears that BootRom 3.1.3 works with vsftpd 2.0.1, so I placed bootrom 3.0.1 (which I know works with vsftpd 2.0.3) on my CentOS server and downgraded the phone to 3.0.1. I then placed 3.0.1 and SIP app 1.6.5 (which I was using the whole time, btw) on my AstLinux server running vsftpd 2.0.3. All was good. So now I am successfully running with the following: Polycom IP 501 Bootrom 3.0.1 SIP 1.6.5 AstLinux 0.3.7 vsftpd 2.0.3 I will also try to fix (or workaround) this by trying the following: upgrading AstLinux to include vsftpd 2.0.4 trying an intermediate BootRom release between 3.0.1 and 3.1.3 (find out exactly where/when it broke) trying an even newer Polycom BootRom when it becomes available upgrading the kernel in AstLinux (I doubt that's it) fiddling with iptables rules in AstLinux (iptables was loaded, but obviously 3.0.1 doesn't have a problem with it) This also might be related to the problems described here: http://forums.digium.com/viewtopic.php? p=14847sid=6e70577c37bd345cfc164a01e64e113a Any thoughts? Comments? Suggestions? P.S. - I will be updating the Polycom config files at http:// www.krisk.org/asterisk/pcom/ to reflect some new changes in this firmware release. I just need to get my phones working first :)! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.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] anonymous caller id causes crash
Hi everybody, this is not directly related to Asterisk, but I'm sure this is the place to get an answer: Even with Asterisk not running, the entire system will crash when a call comes in through CAPI. This only happens when the caller does not submit his caller-id. I'm using the following setup: AMD Athlon 64Bit Open SUSE 10 AVM FritzCard 2.0 PCI Outgoing calls and not anonymous calls are working just fine. I'm grateful for any tip. Regards, Lius ___ --Bandwidth 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: ON DEMAND call Recording
On Mar 7, 2006, at 2:38 AM, Giridhar Bandi wrote: ya i found it it *1 to start recording from the caller end Also pushing *1 again stops recording. ___ --Bandwidth 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] pap2 Dial plan
Hi i am using pap2 phone adaptors as clients to connect to asterisk server i am able to make calls but i cannot access voice mail using phone or start recording while call is in progress and when i place a call to local sip extension there is a long pause ( 15 sec ) before the call gets dialled i assume that the problem would be due to the dial plan in PAP2 if so please help me changing it thanks Giridhar Bandi ___ --Bandwidth 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] indications SIP
7 mar 2006 kl. 17.00 skrev Can2002: Apologies if this is an old question; I've searched the list and the wiki but have not been able to find a definitive answer. I have an Aastra 480i phone registered with * 1.2.4; I want to generate UK ringback tones when the handset dials another internal extension. On my Zap channels, I have this in place by editing /etc/zaptel.conf; however I've had no luck with the Sip handset (I have the same problem with a Grandstream ATA). My indications.conf has country=uk and I've also set both the general and extension sections in sip.conf to language=uk, but I still only get US ringback tones on the Aastra handset. I've probably missed some vital point, but I'd appreciate any pointers people could give. With SIP phones, the phone, not Asterisk, generates all the indications. Check with Aastra. In some cases, like during a call transfer, Asterisk may generate a tone. /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
Re: [Asterisk-Users] Re: ON DEMAND call Recording
Hey thanks for the prompt response ( that's what i liked about this list ) i was not able to start recording i have pap2 box as clients and the dial plan of pap2 is as bellow (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) can you suggest if this is causing the problem thanksGiridhar BandiOn 3/7/06, Martin Joseph [EMAIL PROTECTED] wrote:On Mar 7, 2006, at 2:38 AM, Giridhar Bandi wrote: ya i found it it *1 to start recording from the caller endAlso pushing *1 again stops recording.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] res_mysql.conf DNS SRV lookup
7 mar 2006 kl. 18.12 skrev Douglas Garstang: Are you kidding? Asterisk doesn't do SRV. If you read all the VOIP books out there, SRV lookups are _the_ way to achieve redundancy. Digium hasn't gotten to it I guess with their 'enterprise class' product though. Are you kidding? We do SRV. I dial with it every day. Even if it's broken, we still do SRV. There is work being done to improve the SRV support. I haven't seen it used for MySQL before. What's the SRV record name for this? Any example? /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] Using softphone from a remote location to get into *
HI All, What is a good tutorial or article on using Xlite to get into * while doing so over the Internet? I have had problems with doing this by having one way audio. I had searched around and not found an article that addressed the problem Thanks, -- Leonard Burton, N9URK [EMAIL PROTECTED] The prolonged evacuation would have dramatically affected the survivability of the occupants. ___ --Bandwidth 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] new beta Grandstream firmware HT488_496_386
I have bought more than 20. Maybe 2 of them work well... :-( I have to make cold reset on the ATA_386 every days... Regards Amr _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones Sent: terça-feira, 7 de Março de 2006 13:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 I'm almost afraid to ask, but is the HT 386 known for having a lot of troubles? I just installed one at home about 2 weeks ago, and knock on wood, it's only locked up once, and this was when I was still in the process of tweaking the config to work optimally w/ [EMAIL PROTECTED] I can't say I'm entirely pleased with the slight echo and buzz I'm detecting, but so far it's at least worked.. This isn't the consensus though, huh?! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: sábado, 4 de Março de 2006 0:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 They promised me this for my POS 386 adapters that need to be rebooted every few days from lockups about 4 months ago. Gee I wonder if this will work. Probably not. On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote: http://grandstream.com/BETATEST/HT488_496_386/ attachment: winmail.dat___ --Bandwidth 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] res_mysql.conf DNS SRV lookup
My bad. SRV lookups work, but Asterisk only uses the first entry right? This means there's no redundancy. -Original Message- From: Olle E Johansson [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 07, 2006 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] res_mysql.conf DNS SRV lookup 7 mar 2006 kl. 18.12 skrev Douglas Garstang: Are you kidding? Asterisk doesn't do SRV. If you read all the VOIP books out there, SRV lookups are _the_ way to achieve redundancy. Digium hasn't gotten to it I guess with their 'enterprise class' product though. Are you kidding? We do SRV. I dial with it every day. Even if it's broken, we still do SRV. There is work being done to improve the SRV support. I haven't seen it used for MySQL before. What's the SRV record name for this? Any example? /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 ___ --Bandwidth 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] new beta Grandstream firmware HT488_496_386
Ugh.. That's not good news... I guess I have wither a digium card or a Sipura FXS in my future unless I'm one of the lucky 10% then!! :-) Thanks for the feedback! From: Andre Rodrigues (Cheyenne) [mailto:[EMAIL PROTECTED] Sent: Tue 3/7/2006 1:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 I have bought more than 20. Maybe 2 of them work well... :-( I have to make cold reset on the ATA_386 every days... Regards Amr _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones Sent: terça-feira, 7 de Março de 2006 13:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 I'm almost afraid to ask, but is the HT 386 known for having a lot of troubles? I just installed one at home about 2 weeks ago, and knock on wood, it's only locked up once, and this was when I was still in the process of tweaking the config to work optimally w/ [EMAIL PROTECTED] I can't say I'm entirely pleased with the slight echo and buzz I'm detecting, but so far it's at least worked.. This isn't the consensus though, huh?! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: sábado, 4 de Março de 2006 0:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 They promised me this for my POS 386 adapters that need to be rebooted every few days from lockups about 4 months ago. Gee I wonder if this will work. Probably not. On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote: http://grandstream.com/BETATEST/HT488_496_386/ winmail.dat___ --Bandwidth 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] Using softphone from a remote location to get into *
Is the softphone behind NAT ? If it is insert nat=yes in your dial plan. Is the server behind NAT ? If it is you need to open ports 5060,5061 and 1-2. Dovid --- Leonard Burton [EMAIL PROTECTED] wrote: HI All, What is a good tutorial or article on using Xlite to get into * while doing so over the Internet? I have had problems with doing this by having one way audio. I had searched around and not found an article that addressed the problem Thanks, -- Leonard Burton, N9URK [EMAIL PROTECTED] The prolonged evacuation would have dramatically affected the survivability of the occupants. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Using softphone from a remote location to get into *
take a look at the following things- enable nat on asterisk - if you are using a perimeter firewall then forward port 5060 , 1 -2 ( these are default ) - use correct sip proxy address on you xlite phone --Giridhar Bandi On 3/7/06, Leonard Burton [EMAIL PROTECTED] wrote: HI All,What is a good tutorial or article on using Xlite to get into * whiledoing so over the Internet?I have had problems with doing this by having one way audio.I hadsearched around and not found an article that addressed the problem Thanks,--Leonard Burton, N9URK[EMAIL PROTECTED]The prolonged evacuation would have dramatically affected thesurvivability of the occupants. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Setting Vaaibles
Helo List,
First I would like to apologize for my bad spelling as
well as that I did not search the wiki first. I only
have email access at the moment.
I am having trouble setting both variables and global
variables thru an extension.
I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4
with an Xlite softphone. I have two xlite phones on
diffent computers. One logs in as xlite1 and the other
as SNOM.
My dial plan is as follows
Exten = 200,1,Dial(${OnCall},10)
Exten = 201,1,Set(OnCall=SIP/SNOM)
Exten = 202,1,Set(OnCall=SIP/xlite1)
(I have tried Set and SetGlobalVar).
When I use Set I get the following in the CLI
-- Executing Set(SIP/snom-a645, OnCall=SIP/SNOM)
in new stack
== Auto fallthrough, cahnnel 'SIP/snom\a645 status is
'UNKNOWN'
If I dial ext. 201 or 202 I get call failed: 603
declined on the xlite phone. When I dail 200 I get an
error
If I use SetGlobalVar the output from the CLI is
-- Executing SetGlobalVar(SIP/snom-24f8,
OnCall=SIP/SNOM) in new stack
= Setting global variable 'OnCall' to 'SIP/SNOM'
== Auto fallthrough, channel 'SIP/snom-24f8' status is
'UNKNOWN'
When I use SetGlobalVar I get the same error in the
xlite phone. However when I dial ext. 200 it works.
I tried dialing 201 and 202 from both softphones and I
got the same errors.
Thanks a lot.
Dovid
__
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Re: [Asterisk-Users] I can't receive multiple pages with spandsp
Marco Maiolini wrote: Hi all, I'trying to use spandsp (app_rxfax) to receive faxes. When there are more than one page, the system creates a tiff file with only the first page and the other are lost, even if the full log says: You need a fax viewer that can handle multi-page tif files Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] res_mysql.conf DNS SRV lookup
7 mar 2006 kl. 19.03 skrev Douglas Garstang: My bad. SRV lookups work, but Asterisk only uses the first entry right? This means there's no redundancy. That is correct. That is what we try to fix. /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
RE: [Asterisk-Users] new beta Grandstream firmware HT488_496_386
Yeap... very bad feedback... But I think that the HT 286 model had the same problem, and now they are working well. I will have 3 of them next week to replace the HT 386 models that are using fax lines and working very bad, but consider that the HT 386 hangs a loto f times and I don´t have any clue about this problem... I would prefer the sipura units... (The new ones from cisco!) Regards. Amr _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones Sent: terça-feira, 7 de Março de 2006 18:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 Ugh.. That's not good news... I guess I have wither a digium card or a Sipura FXS in my future unless I'm one of the lucky 10% then!! :-) Thanks for the feedback! _ From: Andre Rodrigues (Cheyenne) [mailto:[EMAIL PROTECTED] Sent: Tue 3/7/2006 1:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 I have bought more than 20. Maybe 2 of them work well... :-( I have to make cold reset on the ATA_386 every days... Regards Amr _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones Sent: terça-feira, 7 de Março de 2006 13:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 I'm almost afraid to ask, but is the HT 386 known for having a lot of troubles? I just installed one at home about 2 weeks ago, and knock on wood, it's only locked up once, and this was when I was still in the process of tweaking the config to work optimally w/ [EMAIL PROTECTED] I can't say I'm entirely pleased with the slight echo and buzz I'm detecting, but so far it's at least worked.. This isn't the consensus though, huh?! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: sábado, 4 de Março de 2006 0:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] new beta Grandstream firmware HT488_496_386 They promised me this for my POS 386 adapters that need to be rebooted every few days from lockups about 4 months ago. Gee I wonder if this will work. Probably not. On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote: http://grandstream.com/BETATEST/HT488_496_386/ attachment: winmail.dat___ --Bandwidth 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] Receiving Multiple calls on asterisk at home
All - I've been muddling around with this for a few days now.. and I'm trying to figure out why I am not receiving more than one phone call on each polycom 501 phone. I can make more than one phone call out, but not receive another one in, while on a call. Has anybody seen this behaivior before, or is there something simple in the config i'm missing, like.. maxcalls.. or something. Thanks! Rolf Brusletto ___ --Bandwidth 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] Changing REINVITE status of the channel dynamically
I've an Asterisk server running in my office, which forwards all
long distance calls to a third party SIP service using an extension rule:
exten = _1XX0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED])
(1XX0 is the international calls rule for Chile)
Also, in my sip.conf, I've defined canreinvite=yes to decrease the
network load to the server caused by the RTP.
However, the external sip server seems to be buggy, because the
REINVITE's against it only works for certain routes, and in others, it
simply hang up the calls. Since I don't have control over that remote
service (and I already inform them about this problem), I'd like to know
if it's possible to set the REINVITE on or off dynamically, based on the
extension being dialed. I don't like very much the option to completely
disable the REINVITE's in my network (formed by a central, and a lot of
offices connected to it by not too fast links, so the network usage is
an issue)
Thanks a lot for your help.
--
Atly.
Alvaro Palma
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Re: [Asterisk-Users] HW Echo Cancellers
Hello, Works great for me as well. over 3 months in production with no problems/no echos. MATT--- On 3/7/06, ADEGOKE ARUNA [EMAIL PROTECTED] wrote: Steve, I just complete a setup of asterisk server in a production environment with a single A104D and there is no echo and the quality is okay. goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: Tuesday, March 07, 2006 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HW Echo Cancellers Hi, In December, I posted an enquiry asking whether anyone had experience with the Sangoma A104d cards (see below) - I got a couple of responses, but basically it was that people have started playing with them, and would publish feedback at a later date. Does anyone have any further feedback at this stage? Many thanks in anticipation. Regards, Steve On 12/20/05, Steve Davies [EMAIL PROTECTED] wrote: http://www.google.com/search?q=cache:3AXi4YvnS80J:www.sangoma.com/company/ne ws_releases/octasic.htmhl=en it seems that there will soon be an A102d, A104d and A108d available on the market. Given that only the A104d is available at present, can anyone give feedback on this product from an asterisk/end-user point of view? Is the EC any good? Does it solve your problems? Are the drivers stable? How is the voice quality? What size of server CPU did you use/need? etc etc etc... Any opinion would be useful to save us investing $$$s at this stage :). ___ --Bandwidth 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] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3
HTTP's nice, but FTP does the job. Check the docs for supported FTP servers -- many of the stock Linux FTP servers will give the exact problem you discussed, below. I should know -- took me almost a week before trying proftpd, and WHAMMO, worked like a champ. -Ken On Tue, March 7, 2006 12:37 pm, William M Conlon wrote: I spent a weekend battling similar issues with 501s, using FC4/ proftpd. I finally switched from FTP to HTTP. On Mar 7, 2006, at 9:53 AM, Kristian Kielhofner wrote: Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated bootrom, nice. Rebooted again. Updated sip firmware. Also nice. Upon the next reboot, the wheels started falling off. The phones would not get changes I made to any of the .cfg files. Several phones would take 20 minutes or more to boot, only to display a 0x4000 config file error. What happened? I have been using various Polycom's with AstLinux (and vsftpd 2.0.3 that I include with it) for quite some time, with no problems whatsoever. Until now. I had been running bootrom 3.0.1 and various versions of the SIP image at several other sites with no problem. At this point I was still unable to accept the fact that I might not be able to run this latest bootrom. After many trial and tribulations, I finally rsync'ed (with -avr) the FTP directory from the AstLinux machine to my laptop running CentOS 4. I configured the vsftpd daemon (version 2.0.1) IDENTICALLY (with the exception of PAM and TCP wrappers) and crossed my fingers... After re-configuring the IP 501 to use my laptop, imagine my surprise when the most problematic of them booted right away without problems. Again and again, everything was fine. So now I just had to break out ethereal and see what was going on. While I have not completely finished my analysis, it appears that Polycom firmware 3.1.3 bombs out when transferring files with vsftpd 2.0.3. The symptom appears to be repeated TCP SYNs from the Polycom to the ftp daemon on port 20. The Polycom will keep retrying and increment its source port number by one every few minutes. Like I said, I need to dig into this more, but I figured I'd report what I know and see if anyone out there can fill in the holes. Here's what I did. It appears that BootRom 3.1.3 works with vsftpd 2.0.1, so I placed bootrom 3.0.1 (which I know works with vsftpd 2.0.3) on my CentOS server and downgraded the phone to 3.0.1. I then placed 3.0.1 and SIP app 1.6.5 (which I was using the whole time, btw) on my AstLinux server running vsftpd 2.0.3. All was good. So now I am successfully running with the following: Polycom IP 501 Bootrom 3.0.1 SIP 1.6.5 AstLinux 0.3.7 vsftpd 2.0.3 I will also try to fix (or workaround) this by trying the following: upgrading AstLinux to include vsftpd 2.0.4 trying an intermediate BootRom release between 3.0.1 and 3.1.3 (find out exactly where/when it broke) trying an even newer Polycom BootRom when it becomes available upgrading the kernel in AstLinux (I doubt that's it) fiddling with iptables rules in AstLinux (iptables was loaded, but obviously 3.0.1 doesn't have a problem with it) This also might be related to the problems described here: http://forums.digium.com/viewtopic.php? p=14847sid=6e70577c37bd345cfc164a01e64e113a Any thoughts? Comments? Suggestions? P.S. - I will be updating the Polycom config files at http:// www.krisk.org/asterisk/pcom/ to reflect some new changes in this firmware release. I just need to get my phones working first :)! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Path Optimization?
Hello, Is Call Path Optimization (IAX Draft, Section 6.4.4) supported by Asterisk? If not, is there a roadmap for it? If there is a URL I can study to get the answer, I will appreciate a pointer. I scanned recent mailing list archives and couldn't find any discussion of this topic. Thank you. Vipul Bhatt, Asterisk-newbie ___ --Bandwidth 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] Send One Touch Record to mail
As far as I know, you will need to do this yourself with some creative scripting. There was some talk on the list awhile ago to move the recording tovoicemail, but I dont' know if anyone has made a patch to do it yet. On 3/7/06, Tomislav Parčina [EMAIL PROTECTED] wrote: How can I send recordings, that I have recorded with One Touch Record, to e-mail address that is defined in voicemail.conf?Thank you for your ideas.--Tomislav Parcinatparcina#lama.hr___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Re: ON DEMAND call Recording
On Mar 7, 2006, at 9:51 AM, Giridhar Bandi wrote: Hey thanks for the prompt response ( that's what i liked about this list ) i was not able to start recording i have pap2 box as clients and the dial plan of pap2 is as bellow (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) can you suggest if this is causing the problem Dunno, did you add the wW in your dial command? and then reload? That worked for me, as I did this yesterday. Also I enabled automon in features.conf. Pretty slick. Marty ___ --Bandwidth 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] Question from a newbie on finding digium hosts
Hey all, I have a client whose previous programmer ditched. I'm his webmaster, and now he wants me to have an asterisk system set up for serial number authentication...and I have a digium card from the previous guy. Are there hosts that will set this up for me? ie, rack space somwhere? Are there guides online I can look at? Thanks Razib ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
