Re: [asterisk-users] Help with Dialplan Rules Please!
Thanks Alex, it looks like you had a great answer to the issue at hand. On 10/17/06, Alex Robar [EMAIL PROTECTED] wrote:If the order is giving you problems, create two separate outbound routes, one for local calls and one for long distance. Make sure the local route is before the LD route, and it should work for you. Both outbound routes can use the same trunk without issue. AlexOn 10/17/06, Chris Ramsey [EMAIL PROTECTED] wrote: This was posted at The Asterisk Blog Forums Click here for the original post. I need someone to explain how the dialplan rules work? I'm having a hard time getting it. I know that to dial out you need a 9 and to ignore that 9 once your out... requires a switch of sorts that tells asterisk to ignore the first digit on the left. In freePBX it's this: 9|NXX For Long distance it is 9|1NXXNXX Here is my problem using Free PBX: I want to be able to dial long distance and local at will while using free PBX to set it up. Right now we have 1 line for testing purposes and soon to be expanded into 2. When the rules are arranged like this in FreePBX 9|1NXXNXX 9|NXX the long distance portion works but the local one does not. When its arranged like this 9|NXX 9|1NXXNXX They both work! But the above is only done when it's hard coded into the configuration file (additional_extensions.conf) and free PBX always puts it in this order... wether I like it or not. 9|1NXXNXX 9|NXX And causes problems in the configuration file when and I change the settings. This isn't going to help me much! Im just a tad bit confused. A Little help here?-- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___ --Bandwidth 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 hardware and is it possible
I want to buy Digium card which one is the best? too many version make me confuse I have T1 connection for this stuff On 10/17/06, Noah Miller [EMAIL PROTECTED] wrote: Hi Ady - Imagine i want to create application like SMS Alert, however it's a call alert when something happened, for example server is crashed, i want to call 100 of my staff (administrator, manager, and others) using asterix, when they pick up their phone, my asterix will play an audio file Is it possible? Yes. For more information: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out what is the correct hardware for this? Any modern Linux, BSD (including OS X), or Solaris compatible computer to run asterisk. If you are using an ITSP (VoIP provider) you don't need any other hardware than your network card. If you have a PSTN phone connection, at the very least you'll need a card (like Digium, Sangoma, Rhino, etc), or an external gateway (like linksys, dlink, mediatrix, etc). - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CAPI channel not available but nobody is using the system
On Tue, 17 Oct 2006, Tim Sharp wrote: I have 23 CAPI channels defined and normally multiple channels are in use during the day for outbound calling. The problem is that every 3 or 4 months one of the channels becomes unavailable and then no calls can come in or go out on any of these channels. CAPI INFO shows Contr1: 23 B channels total, 22 B channels free. To fix the problem I reboot the asterisk server. First, is there a better way to reset the channels than rebooting? It depends where the problem really has its origin. If just asterisk (chan-capi) has a wrong channel count, it would be enough to unload chan-capi. Maybe asterisk itself need to be restarted. But if the real problem comes from the CAPI/ISDN driver, you need to reload these drivers. Which version of asterisk/chan-capi do you use? What ISDN hardware / CAPI driver do you use? Second, is there a way to bypass the unavailable channel in the dialplan? No. Third, what is causing the problem and can I prevent it? chan-capi counts the active channels when the CONNECT/DISCONNECT message of b-channels are indicated. If one of these messages are missing (it's a bug in the CAPI driver if that happens) the count is wrong. Armin ___ --Bandwidth 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] Reception Console
Yes, please. I would love to test for you. Med venlig hilsen / Best regards, Henrik Woffinden Technical Director Nitram Lexa ApS Maglebjergvej 5A DK-2800 Kongens Lyngby Denmark Phone: +45 70 25 24 23 Fax: +45 70 25 29 23 Mobile: +45 40 85 25 17 E-mail: [EMAIL PROTECTED] Web: www.nitramlexa.com --- Windows is a 32-bit extension to a 16-bit graphical shell for an 8-bit operating system originally coded for a 4-bit microprocessor by a 2-bit company that can't stand 1 bit of competition. Paul Hales wrote: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales ___ --Bandwidth 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] Nokia E60/61/70 and SIP
Martin Joseph wrote: For all of us using these devices, I have some good news. There is a self installable firmware update available from Nokia here (requires windows box to install): http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate This seems to radically improve the behavior of the SIP client on my E60. It seems to have resolved several of the MANY bugs that were outstanding on this product. The update does erase all your setups and info though. You are warned. Marty Hi, there are differences for the lang? I have the E60 in italian lang and the software update says that I have the last firmware (I don't think is the last firmware). I have a problem with this phone and Asterisk, my sip.conf is: [208] username=208 type=friend secret=1234 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes (I have try with neven too) notransfer=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Pasqu 208 disallow=all allow=alaw ## I have set my phone with this conf: ## Public user name: sip:[PhoneNumber]@:[youre asterisks servers ip] or sip:[EMAIL PROTECTED] Use Compression: No Registration: Always on or When Needed Use Security: No -- Proxy Server: sip:[youre asterisks servers ip] Realm: asterisk User Name: PhoneNumber Password: PIN Allow loose routing: Yes Transport Type: UDP Port: 5060 -- Registrar Server: sip:[youre asterisks servers ip] Realm: asterisk User Name: PhoneNumber Password: PIN Transport Type: UDP Port: 5060 #3 but the phone not work. In the log I read this: ### with nat = never -- SIP read from 151.38.43.46:19834: REGISTER sip:192.168.1.200 SIP/2.0 Route: sip:192.168.1.200;lr Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKoshj5kfbp1hc74qm0ackh80 From: sip:[EMAIL PROTECTED];tag=jmtj5k9nadhc7fim0ack To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED];expires=3600 CSeq: 1126 REGISTER Call-ID: vfQOQl--oIeXj0api9F6nimvnTwG0T Supported: sec-agree Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.1.99 : 5060 (NAT) Transmitting (no NAT) to 192.168.1.99:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKoshj5kfbp1hc74qm0ackh80;received=151.38.43.46 From: sip:[EMAIL PROTECTED];tag=jmtj5k9nadhc7fim0ack To: sip:[EMAIL PROTECTED] Call-ID: vfQOQl--oIeXj0api9F6nimvnTwG0T CSeq: 1126 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 192.168.1.99:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKoshj5kfbp1hc74qm0ackh80;received=151.38.43.46 From: sip:[EMAIL PROTECTED];tag=jmtj5k9nadhc7fim0ack To: sip:[EMAIL PROTECTED];tag=as29a9cc9f Call-ID: vfQOQl--oIeXj0api9F6nimvnTwG0T CSeq: 1126 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2495e6cb Content-Length: 0 ### with nat = yes # -- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 151.38.43.46:20300: REGISTER sip:192.168.1.200 SIP/2.0 Route: sip:pasqu.zapto.org;lr Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKegpl50ntmlhc69ar0ackhhd From: sip:[EMAIL PROTECTED];tag=5vk550gjc5hc7o4v0ack To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED];expires=3600 CSeq: 1133 REGISTER Call-ID: 5hdiwAD3oIdgxgapi3CfvgbWvzQ0PU Supported: sec-agree Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.1.99 : 5060 (NAT) Transmitting (NAT) to 151.38.43.46:20300: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKegpl50ntmlhc69ar0ackhhd;received=151.38.43.46 From: sip:[EMAIL PROTECTED];tag=5vk550gjc5hc7o4v0ack To: sip:[EMAIL PROTECTED] Call-ID: 5hdiwAD3oIdgxgapi3CfvgbWvzQ0PU CSeq: 1133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 151.38.43.46:20300: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bKegpl50ntmlhc69ar0ackhhd;received=151.38.43.46 From: sip:[EMAIL PROTECTED];tag=5vk550gjc5hc7o4v0ack To: sip:[EMAIL PROTECTED];tag=as6985bfa3 Call-ID: 5hdiwAD3oIdgxgapi3CfvgbWvzQ0PU CSeq: 1133 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest algorithm=MD5,
[asterisk-users] Please explain these SIP errors
Hi, sometimes on by Asterisk 1.2.10 box I get these errors, there are about 50 active SIP channels so I dont know if calls are getting dropped or not. Should I be worried? 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! -- Executing GotoIf(SIP/sipCSC-b737f9e8, 0 ? 15) in new stack 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! == Spawn extension (iax, 0837707300, 34) exited non-zero on 'SIP/sipCSC-b73aba28' 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11347 sipsock_read: We could NOT get the channel lock for SIP/sipCSC-b73aba28! 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11348 sipsock_read: SIP MESSAGE JUST IGNORED: ACK 2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11349 sipsock_read: BAD! BAD! BAD! == Spawn extension (iax, 0825905581, 24) exited non-zero on 'SIP/sipBBG-b736f910' -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth 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] Inaccurate CDRs
I have found the problem. Before calls leave our network, thee user must supply a pin. this is a for of call accounting that we implemented. To do this, we had used AMP's Authenticate () function. This function actually and always answers the channel first before accepting pin entries. This was why there is always an answered flag on the channel. and since the channel is answered as soon as the call is made, there is no difference between the duration and the billsec. Now my problem is how do i implement an authentication AGI that uses DTMF ? i would be posting this question in another thread Thanks for your help On 10/17/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: this Cdr Record if from the Primary PBX '2006-10-17 07:11:37', 'Admin', 'XXX, 'aa', 'from-internal', 'IAX2/[EMAIL PROTECTED]' , 'Zap/1-1', 'ResetCDR', 'w', 10, 0, 'BUSY', 3, '', '', '' this is the CDR record from the secondsry for the same call '2006-10-17 13:31:57', 'Admin X', 'X', 'aa', 'from-internal', 'SIP/401-8f0c', 'IAX2/TRUNK1-2', 'Dial', 'IAX2/TRUNK1/aaa|120', 15, 15, 'ANSWERED', 3, '4147', '', '' in this setup, the caller dropped the call after allowing it to ring for 15 seconds On 10/17/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: Well I am using APM on the two boxes i have modified the srripts extensievely and i am sure that there is no Awnser befor a dial when Dialing through the PBX trunks On 10/17/06, Steve Davies [EMAIL PROTECTED] wrote: On 10/17/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls ( i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my call disposition colume. I have also noticed that both the call duration and the billsec are always equal. here is my setup below PSTN va E1 (Primary Asterisk) =Sip and IAX trunks (Secondary PBX) Clients are connected to the Secondary PBX. this pbx handles registration of all clents. The billing irregularities happen on the Secondary PBX. When a call is maked from the Secondary and it is routed across the trunks, call disposition always registeres 'AWNSERED', unless the Primary PBX sends back a busy signal. the duration and billsecs are always equla. this means that the user gets billed for ring time, and calls disconnected from the Secondary PBXCould you provide a snippet of the dialplan used on each of theprimary and secondary boxes to complete a call?For example, is the primary executing an Answer() before it does the onward Dial() on behalf of the secondary?Cheers,Steve___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Is 1.2.12.1 production ready (Mauro Zanin)
Hi Everybody, as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but VoiceMail application stated that there were no entries in voicemail.conf, so it didn't work. Installed again 1.2.0 and voilà the VoiceMail app. was working again. I asked to the group, but it seems I'm the only one with this issue! In Italy we say: Chi lascia la via vecchia per la nuova, sa quel che perde e non sa quel che trova litterally: Who leaves the old way, does know what he loses and doesn't know what he finds. Ciao Mauro ___ --Bandwidth 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] AGI for authenticating calls with DTMF
HI, I am trying to write an AGI that will authenticate users befoer the call is allowed to proceed. Befor you ask, i have tried using the authenticate() function but it does not work for me as this function messes up call accounting (authenticat() awayas awnseres the channel, thus causes CDR to bill for 'ring' time). the AGI will (1) playback a voice prompt over an unawnsered channel (2) Read DTMF input form keypad (3) use this numbers to authenticate/validate the numbers entered from a pin-code list (4) disconnect or complete the call based on the validity of the pin code entered and confirm results with voice prompt Any helpful resources would be appriciated. ___ --Bandwidth 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] Cisco 2621 NM-HDV VWIC-1MFT1
http://www.voip-info.org/wiki/index.php?page=Asterisk+cisco+FXO is a
good read for that.
I've got a couple of 2600's configured this way, and all seems to work
just fine. One little detail I came across was one-way-audio..
strangely enough that was fixed if I used
Dial(SIP/${EXTEN:[EMAIL PROTECTED],40,to) .. the o Dial-option fixed
it in my dialplan, both for outgoing and incoming calls.
SIP calls from the 2600 arrive in your asterisk in the form
[EMAIL PROTECTED], my approach was to let it use default context, then
match the numbers there with exten and send it off to the individual
contexts from there with Gosub().
Good luck :-)
On 10/18/06, David Edwards [EMAIL PROTECTED] wrote:
Steve,
I was just looking for a little info to get me started..
Thanks
David
- Original Message -
From: Steve Blair [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 17, 2006 19:24
Subject: Re: [asterisk-users] Cisco 2621 NM-HDV VWIC-1MFT1
David:
Do you have a specific problem with this card? If not and you are just
looking for general information you can try the following document.
-Steve
http://mit.edu/sip/sip.edu/ciscoGW.html
David Edwards wrote:
Hi all,
We are trying to use a Cisco 2621 with NM-HDV VWIC-1MFT1 to connect
to a PBX via the PRI card. We want to use it as a gateway to forward
all calls to a hosted Asterisk server off-site via SIP.
Does any one have any suggestions on how to best approach this?
Thanks
David
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[asterisk-users] Netgear WGT Flash-fest at Astricon
Just an FYI to anyone out there who will be attending Astricon and who would like to play around with embedded Asterisk on the Netgear WGT634U platform. If you want to bring your own to the show, I'll be bringing all the appropriate stuff to flash them there with my latest openWGT/Asterisk build. They are available from www.justdeals.com, refurbs, for $44.95 delivered. You'll also need a USB flash drive. I use 256MB, but Asterisk can be set up to use as little as 32MB. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth 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: 1.4 on mac OSX 10.4.8
On 2006-10-17 14:19:00 -0700, Daniel Salama [EMAIL PROTECTED] said: You can get wget for OSX from DarwinPorts (http://wget.darwinports.com/) Ok, I bit the bullet and build wget. This allows me to build 1.4 branch, which does the same thing as 1.40b2. It starts up, consumes as much CPU as is available, and is not responsive to CLI commands or registrations. It's a dead duck. Anybody out there trying this stuff on OSX? 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
Re: [asterisk-users] Reception Console
Hi all Looks like the person not responding all does one got URL for testing from him ? Ram On 10/18/06, Henrik Woffinden [EMAIL PROTECTED] wrote: Yes, please.I would love to test for you.Med venlig hilsen / Best regards,Henrik Woffinden Technical DirectorNitram Lexa ApSMaglebjergvej 5ADK-2800 Kongens LyngbyDenmarkPhone: +45 70 25 24 23 Fax: +45 70 25 29 23Mobile: +45 40 85 25 17E-mail: [EMAIL PROTECTED] Web: www.nitramlexa.com---Windows is a 32-bit extension to a 16-bit graphical shell for an 8-bitoperating system originally coded for a 4-bit microprocessor by a 2-bit company that can't stand 1 bit of competition.Paul Hales wrote: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales___--Bandwidth 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] IAX2 thru NAT problem
Hi people, i have problem with IAX2 between two asterisk PBX. When i try call some number i get INVAL packet, but when i try call same number via OpenVPN (is between this two asterisk) call is working fine.So i debug communications and here is my opinion ... Schema of connection: Asterisk1 - ADSL router with NAT - INTERNET - Asterisk2 A)Calling directly via public IP's (port 4569 is forwarded on ADSL modem to asterisk1) - not working Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 4 DCall: 0 [213.160.177.186:4569] VERSION : 2 CALLED NUMBER : 1299 CODEC_PREFS : () CALLING NUMBER : 1199 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Marian_Rychtecky LANGUAGE: en USERNAME: some_username FORMAT : 2 CAPABILITY : 2097151 ADSICPE : 2 DATE TIME : 2006-10-18 10:16:14 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 6ms SCall: 3 DCall: 4 [213.160.177.186:9785] AUTHMETHODS : 3 CHALLENGE : 585590037 USERNAME: VALSABBIA-SLOVENSKO Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 3 [213.160.177.186:9785] B) calling thru openvpn - working Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 4ms SCall: 1 DCall: 0 [192.168.255.2:4569] VERSION : 2 CALLED NUMBER : 1299 CODEC_PREFS : () CALLING NUMBER : 1199 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: Marian_Rychtecky LANGUAGE: en USERNAME: user_name FORMAT : 2 CAPABILITY : 2097151 ADSICPE : 2 DATE TIME : 2006-10-18 10:14:16 -- Called VALSABBIA-SLOVENSKO:[EMAIL PROTECTED]/1299 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00012ms SCall: 1 DCall: 1 [192.168.255.2:4569] AUTHMETHODS : 3 CHALLENGE : 186694617 USERNAME: VALSABBIA-SLOVENSKO Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00034ms SCall: 1 DCall: 1 [192.168.255.2:4569] MD5 RESULT : b0674601456416db7e474de9a858c742 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00041ms SCall: 1 DCall: 1 [192.168.255.2:4569] FORMAT : 2 Only difference what i see is that in first case is the source port of far-end changed from 4569 to 9785 because of NAT of ADSL modem.In case of calling thru openvpn is port unchanged ... It is possible thats the problem? Can somebody help me with my problem? Thanks a lot -- Marian Rychtecky [EMAIL PROTECTED] Tel. +420 724 397 441 ICQ 76582857 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium on Dell PowerEdge 1850
Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Best regards, -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.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] Server power indication
Hello list, I'm currently looking into building a new Asterisk server, due to some codec problems i've got to transcode most of my channels between Alaw -- G729. Is there any indication on how many channels you would be able to transcode on a certain platform? I'm looking into dual Xeon or dual Opteron configurations, which of these platforms would perform better? And how much power would be needed to transcode 120 Channels PRI to G729 (for example Digium TE412P)? Erik ___ --Bandwidth 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] Blank page when sending faxes (repost)
Guys, [Forgive this repost -- my original posting doesn't seem to have shown up in the list (yet)] I have been trying to send a fax using spandsp/app_rtxfax/asterisk. I'm trying to send a (3 page) fax, and all I'm receiving is a single blank page (and I've tried it with 2 fax machines). As some background, I'm using asterisk with an E1 connection through a Sangoma WAN card -- all of this has been working with voice for over 6 months. The software is all working too from what I can see, I've got tiff 3.8.2, app_rtxfax 0.0.2_pre25 and spandsp-0.0.2_pre26 with asterisk-1.2.9_p1 on Gentoo Linux. I have created a simple test using ghostscript to create a TIFF image in the correct format. I've enabled debugging in spandsp (and have the caller argument, which seems to be a common pitfall), and this is what I'm seeing: Restarting V.29 Restarting V.27ter Changed from phase 0 to 2 HDLC carrier up HDLC carrier down HDLC carrier up HDLC carrier down HDLC carrier up HDLC carrier down HDLC carrier up HDLC framing OK Changed from phase 2 to 3 HDLC carrier up HDLC framing OK HDLC carrier down HDLC carrier up HDLC framing OK NSF without final frame tag The remote was made by 'HP' CSI without final frame tag Remote fax gave CSI as: DIS with final frame tag In state 10 ???: 3rd generation mobile network V.8 capable Prefer 64 octet blocks Reserved: 0x98 Supported data signalling rates: V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = T3.85 Uncompressed mode Reserved: 0x10 Minimum scan line time for higher resolutions: T15.4 = T7.7 ???: Prefer 256 octet blocks Reserved: 0x80 Supported data signalling rates: V.27ter fallback mode 2D coding Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 20ms at 3.85 l/mm: T7.7 = T3.85 Start sending document Start tx document Changed from phase 3 to 4 HDLC underflow in state 3 Restarting V.29 Changed from phase 4 to 6 Changed from phase 6 to 3 HDLC carrier up HDLC framing OK CFR with final frame tag In state 4 Trainability test succeeded Start tx page 0 HDLC carrier down Restarting V.29 Changed from phase 3 to 6 Changed from phase 6 to 4 Start tx page 1 HDLC underflow in state 13 Changed from phase 4 to 3 HDLC carrier up HDLC framing OK RTN with final frame tag In state 13 Changed from phase 3 to 4 HDLC underflow in state 2 Disconnecting Changed from phase 4 to 7 Changed from phase 7 to 8 From what I can see, this looks plausible, and certainly shows that there is a significant conversation going on. The Fax Activity Log shows the comany name that I have set in the software, and 'OK' as the result, and yet only a blank page was produced! I've also created another TIFF image using 'pnmtotiff -g3' and this has behaved in the same way. Any insight will be gratefully accepted, Regards, Nick Glencross ___ --Bandwidth 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: Blank page when sending faxes (repost)
On 18/10/06, Nick Glencross [EMAIL PROTECTED] wrote: Guys, [Forgive this repost -- my original posting doesn't seem to have shown up in the list (yet)] Just my luck, my original posting has just appeared. Sorry for the noise! Nick ___ --Bandwidth 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] ISDN PRI and R2 Trunks in one Dual Port Card
On Tue, Oct 17, 2006 at 06:14:43PM -0700, Angel Heart wrote: Hi Guys, Anyone can tell where can I look all your previous post, I am wondering what could my zapata.conf be if I wanted to use two(2) different Trunk Protocol (ISDN R2) in a single Dual Port Digium Card. Sorry, I'm a new user in this forum and new asterisk user as well. Hope somebody could lend a hand/knowledge about this set-up. For R2 use chan_unicall . It is not supported directly with chan_zap. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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] Remote UNIX connection, Remote UNIX disconnected displayed every second
On Tue, Oct 17, 2006 at 04:08:37PM +0200, Giorgio Incantalupo wrote: Hi Tzafrir, 1) it is provided with asterisk 2) it is called by asterisk init script by default (and the asterisk init script is provided by default too) so I think/hope it is good enough 3) actually I have got nothing else and time is very short s/safe_asterisk/asterisk/ . Just run 'asterisk' , and you're done. It is actually less buggy as it has a proper pid file. Currently the Debian init.d script is buggy for the safe_asterisk case for that point. There's an open bug which I have no idea how to handle. Not to mention that safe_asterisk assumes for some reason that you have a certain tty open (console). Trying to adapt it to run with screen is a pain. And why do you need the asterisk console in the first place when you have the logs and asterisk -r? Why generate the extra burden of verbose logging in the common case? 4) I need something to restart asterisk in case of failure In some cases it will cause more harm. It may cause frequent restarts. And then again: how do you stop it? The script is horribly buggy 5) many people on internet say to use it One person on the internet says not to use it. I used to launch safe_asterisk directly...maybe this was my error...now If you launch asterisk manually twice, you won't get a problem. It will report you that it is already running. I use the init script inside contrib/init.d...maybe I'll be more lucky. Giorgio Incantalupo Tzafrir Cohen wrote: On Tue, Oct 17, 2006 at 10:59:33AM +0200, Giorgio Incantalupo wrote: Hi Brian, yes, I have more copies of safe_asterisk running, I know this is the underline problem but I do not how to solve it because I do not know how to reproduce it. I'm still looking the safe_asterisk for some strange but found nothing till now. Have you got the same problem? Why is it happening? Which brings up the obvious question: why do you need safe_asterisk in the first place? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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] 3way calling / codec problem
Mr. Jones wrote: Is there some way I can tell? On 10/16/06, Thomas Kenyon [EMAIL PROTECTED] wrote: Mr. Jones wrote: I'm having problems with conference calls (3-way) when I have my codec forced to g729 in sip.conf. I'm using Grandstream 2000s. If enable both g711 and g729 then 3 way calling and transfers work. I'm not sure why this would matter? Here's the error: Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs! Any help is greatly appreciated! Are you out of licences? From memory when in a console each channel needs to be able to be transcoded to SLIN. (where it is mixed and transcoded back again). I meant conference (not console). You can show g729 or have a console open with verbosity set (probably to 3) and it should tell you on the console output (usually several times). ___ --Bandwidth 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: Is 1.2.12.1 production ready (Mauro Zanin)
Mauro Zanin wrote: Hi Everybody, as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but VoiceMail application stated that there were no entries in voicemail.conf, so it didn't work. Installed again 1.2.0 and voilà the VoiceMail app. was working again. I asked to the group, but it seems I'm the only one with this issue! In Italy we say: Chi lascia la via vecchia per la nuova, sa quel che perde e non sa quel che trova litterally: Who leaves the old way, does know what he loses and doesn't know what he finds. Ciao Mauro I've not had this problem, but I can say that with 1.2.12.1 if I use chanspy, when the spying handset hangs up asterisk segfaults (kicking all connected calls off). Finding that out was embrassing. (was on production server). ___ --Bandwidth 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] Electric usage of a tdm400p
On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote: Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? Erick, Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN ~5). This translates to 2.7 watts. Assuming a DC/DC converter efficiency of 38% (probably low), you would need about 3.7 watts, per FXS module. About 15 watts, total. What is the TDM card installed in and is a disk drive cable available? Bob... ___ --Bandwidth 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] Orange Flash Light Mitel 5215 - Asterisk - working !
Hi guys, I'm trying to reuse Mitel 5215 from proprietary system now into Asterisk :) ! I've them already with SIP and handling calls sucessfully! I've followed instructions from: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+Mitel+5220 Additional Servers * Outbound Server: Off * Outbound Server URL: blank * Outbound Server Port: blank * Voice Mail Server: mailbox # @ * hostname or IP * Number of rings: 4 * Port: 5060 * Backup Server Timeout: 4 Seconds This is just great, but i still with a problem! The Orange Flash light from those phones stills turning Flashing (ON/OFF) all day. It seems to me could be bad configuration of voicemail server parameter: Voice Mail Server: mailbox # @ * hostname or IP Could you explain me what it means # here ? As the Message button also works to retrieve voicemail messages, i thought to put it like [EMAIL PROTECTED] This works good to retrieve the voicemail pressing message button, but the Orange light keeps turning on and off all day:( Any one can help me on this or has experience with this? Could be a bad interpretation from me about the instructions on wiki. Thanks, 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] Digium Single Span Card Installation
Hi I put in a single span Digium card into my Asterisk box and followed some guidelines I found. The ztcfg -vv command gives the following output. Is this correct and does this mean that Asterisk has recognised my E1 card and I will be able to connect my E1 line and use the 30 channels to communicate outside the Asterisk box? Thanks a million Best wishes Iyer Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 31: Clear channel (Default) (Slaves: 31) 18 channels configured. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium Single Span Card Installation
Hi Also tried the following command and got the following lines. Why is that /proc/zaptel/1 reports all the 31 lines whereas ztcfg reports 18 channels? What do they mean? Thanks Best wishes Iyer # cat /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS/CRC4 1 WCT1/0/1 Clear 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear 9 WCT1/0/9 Clear 10 WCT1/0/10 Clear 11 WCT1/0/11 Clear 12 WCT1/0/12 Clear 13 WCT1/0/13 Clear 14 WCT1/0/14 Clear 15 WCT1/0/15 Clear 16 WCT1/0/16 HDLCFCS 17 WCT1/0/17 Clear 18 WCT1/0/18 19 WCT1/0/19 20 WCT1/0/20 21 WCT1/0/21 22 WCT1/0/22 23 WCT1/0/23 24 WCT1/0/24 25 WCT1/0/25 26 WCT1/0/26 27 WCT1/0/27 28 WCT1/0/28 29 WCT1/0/29 30 WCT1/0/30 31 WCT1/0/31 Clear -Original Message- From: K Y Iyer Sent: Wednesday, October 18, 2006 4:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Digium Single Span Card Installation Hi I put in a single span Digium card into my Asterisk box and followed some guidelines I found. The ztcfg -vv command gives the following output. Is this correct and does this mean that Asterisk has recognised my E1 card and I will be able to connect my E1 line and use the 30 channels to communicate outside the Asterisk box? Thanks a million Best wishes Iyer Zaptel Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 31: Clear channel (Default) (Slaves: 31) 18 channels configured. ___ --Bandwidth 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] cut ip adress from caller id number display (ci$co 7941)
I'm playing with phone ci$co 7941 with sip image (8.02SR1), strange is, that phone displays caller id number with ip address of asterisk server like [EMAIL PROTECTED] I think, this is some bug in firmware, but I would like to find some workaround, maybe using SIP_HEADER function, but seems, that this can be used only when calling from SIP to SIP, i.e. not possible to use SIP_HEADER function when I call SIP phone from IAX channel: Oct 18 12:34:22 WARNING[13242]: chan_sip.c:9307 func_header_read: This function can only be used on SIP channels. any other idea, or tip to another firmware with correct behaviour? thanks PJ ___ --Bandwidth 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] Locking phones at night...
On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote: On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote: On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB Sure is, but files in the filesystem are easier to process from external (non-asterisk) programs. In my case, I have a web interface that locks/unlocks phones too. I find it most convenient to use 'ls' to look up the current status of stuff. asterisk -rx could also be used. Or a phone menu. Problems with a phone menu: how can you tell the status? Obviously for performance and elegance the astdb is superior. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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] how to activate recording (automon)
Thanks for reply. I found the problem (problems , were 2 ) first, I wrongly entered vW instead of wW ... When I fixed it, with snom 320 was Ok, but with at-320 wasn't. Then I configured AT-320 to use rfc2833 instead of inband audio, and so also on at-320 was OK Previously I had to enter inband audio otherwise some remote services using DTMF doesn't work. I think the problem is in the at-320, becouse snom phonee using rfc are OK with that services. Andrea Henry.L.Coleman [EMAIL PROTECTED] ip-pbx.ca To Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 17/10/2006 14.05 Re: [asterisk-users] how to activate recording (automon) Please respond to [EMAIL PROTECTED] p-pbx.ca; Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Hi Andrea, Try the following: featuredigittimeout=1500 ; Slow down digits for the record [featuremap] automon = *0 ; One Touch Record Use both option switches(wW) Check that the dial plan on your SIP phones doesn't preclude this feature code. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi all, If I activate recording for an extension everything is OK. but If I activate call recording on demand i am non able to start recording In principle I should have to press *1, as indictaed in features.conf (I am using almost last asterisk code, updated 2 days ago from svn, version SVN-branch-1.2-r39379M ) Actually it produce no effect at all I am using FreePBX interface, and I saw under General Setting two fields, denoted Asterisk Dial command options and Asterisk Outbound Dial command options Here the help says something about w and W options, but every combination of this options does not produce anything Anyway, apart from FreePBX, what I have to check ? And moreover, what are the correct actions to do to record a call ? Let's say extension 555 calls extension 567, 567 answers the call and then press *1 and no other key ? I am trying with at320 sip phones and snom 320 sip phones thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth 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] nat auto detect ?
Eric ManxPower Wieling wrote: Benjamin Jacob wrote: Hello ppl, This post is to do with the variables 'nat' or 'canreinvite' for sip entities. Idealy users, wont be static, they could be roaming all over the globe. So, setting someone as behind NAT, and disabling canreinvite, etc., restricts the roaming capabilities of a user. No. Almost all devices work fine with nat=yes, even if they are not behind NAT. ___ hmm.. ok..let me rephrase my subject, it shud be canreinvite auto detect? The issue is to set canreinvite to yes or no. In an ideal world, the server shud detect, if it should have media passing thru itself, or allow a peer-to-peer audio flow. Ofcourz this behaviour should be controllable. So, the question is, wot do I set canreinvite to?If two users, who are behind two different NATs, and some beautiful morning, step out into the internet, and then make calls, it would be wonderful to let the audio flow between each other directly, thereby offloading the traffic off the *. Any chances? cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Single Span Card Installation
On Wed, Oct 18, 2006 at 04:28:16PM +0530, K Y Iyer wrote: Hi Also tried the following command and got the following lines. Why is that /proc/zaptel/1 reports all the 31 lines whereas ztcfg reports 18 channels? What do they mean? ztcfg reports the channels you are about to try to configure and the result of that configuration. It does not scan your system for channels. Your /etc/zaptel.conf does not include all the bchan-nnels, and thus not all of them are reported and not all were configured. selfpromo If you want to scan your system for channels and attempt to create a wonrking configuration (one that at least be able to pass ztcfg and run asterisk with), use xpp/utils/genzaptelconf. /selfpromo -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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] Locking phones at night...
On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote: On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote: On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote: On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB Sure is, but files in the filesystem are easier to process from external (non-asterisk) programs. In my case, I have a web interface that locks/unlocks phones too. I find it most convenient to use 'ls' to look up the current status of stuff. asterisk -rx could also be used. Or a phone menu. Problems with a phone menu: how can you tell the status? asterisk -rx requires access to the asterisk console which throws its own bunch of problems with permissions and scalability. I'd then prefer to code it through the manager interface but that seems like a terrible overkill here ;) How would you use a phone menu for that? That sounds interesting. Our users here like doing phonestuff on their phones rather than on websites etc. Conrad ___ --Bandwidth 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] random one way audio and noise between SIP phones on same LAN
Hi, sometimes I have one way calls and noise between sip phones connected to the same LAN so no nat/firewall is involved. I tried with different sip phone models soft phones and the result is the same. I searched inside every log file but found nothing. I made different PBX with different hardware but the result is always the same. Is there anybody experiencing this terrible problem? Considering to monitor a remote PBX via ssh, which text-only application could I use? TIA Giorgio Incantalupo ___ --Bandwidth 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] gotoiftime and Macro question
Is there a way to run a macro in a GotoIfTime statement ?? from the wiki documentation it seems not, but.. I would like to do something like this: . 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?Macro(exten-vm,novm,567)) it does not work, as expected from documentation any workaround to call an extension WITHOUT vm (also if vm for that extension is present...) as a consequence of a Time condition? thanks in advance Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth 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] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M
Hi (Armin)! Does someone knows how to identify the type of the card? The delivery note says it is a V-4BRI-8M, whereas lspci reports a 4BRI-8M. What is it really? Are there any Eicon tools to identify the card type? thanks klaus :0a:03.0 Network controller: Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 (rev 01) Subsystem: Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32, Cache Line Size: 0x10 (64 bytes) Interrupt: pin A routed to IRQ 77 Region 0: Memory at fdeffc00 (32-bit, non-prefetchable) [size=256] Region 1: I/O ports at cc00 [size=256] Region 2: Memory at fc00 (32-bit, non-prefetchable) [size=16M] Region 3: Memory at fdee (32-bit, non-prefetchable) [size=64K] Capabilities: [40] Power Management version 1 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [48] #06 [0080] Capabilities: [4c] Vital Product Data ___ --Bandwidth 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] Locking phones at night...
Conrad Wood wrote: On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote: On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote: On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote: On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB Sure is, but files in the filesystem are easier to process from external (non-asterisk) programs. In my case, I have a web interface that locks/unlocks phones too. I find it most convenient to use 'ls' to look up the current status of stuff. asterisk -rx could also be used. Or a phone menu. Problems with a phone menu: how can you tell the status? asterisk -rx requires access to the asterisk console which throws its own bunch of problems with permissions and scalability. I'd then prefer to code it through the manager interface but that seems like a terrible overkill here ;) How would you use a phone menu for that? That sounds interesting. Our users here like doing phonestuff on their phones rather than on websites etc. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users am I missing something important over here?? DB, more specificaly, having ODBCput and ODBCget operations solve all these issues, dont they. read the post abt Stopping putgoing calls after working hours (well.. the subject says so!! ) have your astdb in sql. simple. create extensions to lock/unlock phones or even check status using astdb in sql. very easy to add/view/modify from a webpage too. or... again.. am i missing something over here? - Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Locking phones at night...
On Wed, Oct 18, 2006 at 12:40:38PM +0100, Conrad Wood wrote: On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote: On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote: On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote: On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB Sure is, but files in the filesystem are easier to process from external (non-asterisk) programs. In my case, I have a web interface that locks/unlocks phones too. I find it most convenient to use 'ls' to look up the current status of stuff. asterisk -rx could also be used. Or a phone menu. Problems with a phone menu: how can you tell the status? asterisk -rx requires access to the asterisk console It requires access to the asterisk control socket: /var/run/asterisk.ctl or /var/run/asterisk/asterisk.ctl, depends on your installation. Check the docs on asterisk.conf on setting it to a different ownership that root.root . which throws its own bunch of problems with permissions and scalability. I'd then prefer to code it through the manager interface but that seems like a terrible overkill here ;) How would you use a phone menu for that? That sounds interesting. Our users here like doing phonestuff on their phones rather than on websites etc. DbGet/DbPut or whatever in the dialplan? (After all, a phone menu / IVR is basically a set of Asterisk contetxts calling each other) -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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] Locking phones at night...
On Wed, Oct 18, 2006 at 05:26:49PM +0530, Benjamin Jacob wrote: Conrad Wood wrote: On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote: On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote: On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote: On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB Sure is, but files in the filesystem are easier to process from external (non-asterisk) programs. In my case, I have a web interface that locks/unlocks phones too. I find it most convenient to use 'ls' to look up the current status of stuff. asterisk -rx could also be used. Or a phone menu. Problems with a phone menu: how can you tell the status? asterisk -rx requires access to the asterisk console which throws its own bunch of problems with permissions and scalability. I'd then prefer to code it through the manager interface but that seems like a terrible overkill here ;) How would you use a phone menu for that? That sounds interesting. Our users here like doing phonestuff on their phones rather than on websites etc. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users am I missing something important over here?? DB, more specificaly, having ODBCput and ODBCget operations solve all these issues, dont they. read the post abt Stopping putgoing calls after working hours (well.. the subject says so!! ) have your astdb in sql. simple. This means that there is a ODBC lookup per call. And if the remote database fails, the PBX fails as well. For the sake of simplicity, it might be preferred to use the internal Asterisk DB. Is there a simple and safe way to query the astdb database outside of Asterisk? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.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] gotoiftime and Macro question
On Wed, 2006-10-18 at 13:39 +0200, [EMAIL PROTECTED] wrote:
Is there a way to run a macro in a GotoIfTime statement ??
from the wiki documentation it seems not, but..
I would like to do something like this:
.
554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?Macro(exten-vm,novm,567))
it does not work, as expected from documentation
any workaround to call an extension WITHOUT vm (also if vm for that
extension is present...) as a consequence of a Time condition?
I presume you can't move the GotoIfTime into the macro itself?
Would something like that work for you? (You might need to doublecheck
the exact syntax!)
exten = 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?novm,567,1)
...
[novm]
exten = _X.,1,Macro(exten-vm,novm,${EXTEN})
...
Otherwise, can you post the rest of your dialplan and/or describe in
more detail what you're trying to achieve?
Conrad
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Re: [asterisk-users] how to config chanspy
On 10/17/06, Thirumal Saminathan [EMAIL PROTECTED] wrote: hi all, please any one help me ,how to configure chanspy application . and also send me if u have any sample configure file. -thiruHi,It could be very simple, like:exten = 123,1,ChanSpy(); Spy all channelsor more accuracy:exten =124,1,ChanSpy(SIP); Spy all sip channels if I can help you more, let me know!-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --Bandwidth 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+SER help
Hi Friends,I want to setup multiple SIP accounts. How can I do this? I have installed Asterisk, created Asterisk SIP extensions and registered in www.sipgate.co.uk. Now, what I have to do? 1) Am I need to install SER or OpenSER in my server along with Asterisk?2) If yes, can you please recommond SER or OpenSER?3) I searched in Internet. But, I didn't find good tutorial for this. Can you please tell me a good link for this? Looking forward to your response. Thank you.Regards,Chandra. Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --Bandwidth 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] Windows and file shares
Message: 12 Date: Tue, 17 Oct 2006 18:07:04 -0700 (PDT) From: sdgesa gaeharth [EMAIL PROTECTED] Subject: Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 None of these steps have made a difference. Any other suggestions? Here is my original post: Can anyone help me to figure out why I can not write to a public share? I was able to join the domain without a problem. I can access the share from an xp box. I just can not write: Access denied. thanks On Windows check two sets of permissions: The share permission and the file permission. By default on a Windows 2003 server the share permission is set to read only. You need to change it to read/write. Even if you have the file permission set to full control for everyone - the share permission of read only will block a user from changing a file. Windows 2000 or older did not default to read only on the share but it would be worth looking at if that is the OS you are using. Paul Gaffney LANStatus, LLC ___ --Bandwidth 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] Locking phones at night...
Tzafrir Cohen wrote: On Wed, Oct 18, 2006 at 05:26:49PM +0530, Benjamin Jacob wrote: Conrad Wood wrote: On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote: On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote: On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote: On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB Sure is, but files in the filesystem are easier to process from external (non-asterisk) programs. In my case, I have a web interface that locks/unlocks phones too. I find it most convenient to use 'ls' to look up the current status of stuff. asterisk -rx could also be used. Or a phone menu. Problems with a phone menu: how can you tell the status? asterisk -rx requires access to the asterisk console which throws its own bunch of problems with permissions and scalability. I'd then prefer to code it through the manager interface but that seems like a terrible overkill here ;) How would you use a phone menu for that? That sounds interesting. Our users here like doing phonestuff on their phones rather than on websites etc. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users am I missing something important over here?? DB, more specificaly, having ODBCput and ODBCget operations solve all these issues, dont they. read the post abt Stopping putgoing calls after working hours (well.. the subject says so!! ) have your astdb in sql. simple. This means that there is a ODBC lookup per call. well.. i believe it wud be beter than running an external script, thru asterisk for every call. gotta test that :-) u'll be eating up cpu, along with asterisk doing its own work. there was some talk abt local dbs n remote dbs and the performance on some voip-info page for asterisk. Cant seem to find it right now. And if the remote database fails, the PBX fails as well. well. thats where redundancy n HA come into picture... for that sake, even the internal Berkely DB could fail. For the sake of simplicity, it might be preferred to use the internal Asterisk DB. aahh.. i wasnt talking of simplistic setups :-) Is there a simple and safe way to query the astdb database outside of Asterisk? as i said. ODBC ops!! cheerz - Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] random one way audio and noise between SIP phones on same LAN
Do you use canreinvite (sip.conf)? Change the setting (setting canreinvite=yes may cause nat problems) nad verify if the problem still appears. Using htis setting you can find out if the Audio problem occurs only when media is relayed via Asterisk (-the problem is caused by Asterisk) or in all cases (the problem is not caused by Asterisk) regards klaus Giorgio Incantalupo wrote: Hi, sometimes I have one way calls and noise between sip phones connected to the same LAN so no nat/firewall is involved. I tried with different sip phone models soft phones and the result is the same. I searched inside every log file but found nothing. I made different PBX with different hardware but the result is always the same. Is there anybody experiencing this terrible problem? Considering to monitor a remote PBX via ssh, which text-only application could I use? TIA Giorgio Incantalupo ___ --Bandwidth 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] Orange Flash Light Mitel 5215 - Asterisk - working !
On Wed, Oct 18, 2006 at 11:43:41AM +0100, Marco Mouta wrote: from those phones stills turning Flashing (ON/OFF) all day. It seems to me could be bad configuration of voicemail server parameter: Voice Mail Server: mailbox # @ * hostname or IP Could you explain me what it means # here ? I think # here is just a shorthand for number, and * is short for asterisk I can see how using those symbols in this context is pretty confusing - perhaps you could correct the entry on the Wiki. Regards, Brian. ___ --Bandwidth 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] Why is this happening?
In the case of you example the IAX2 registration came in from the source port on the far device of 1207. Connections don't just move between ports. I understand all this. However, here is my question. MY on 4569 OTHER SIDE 1027. Is both the incoming and outgoing traffic on OTHER SIDE going in and out of 1027? I understand IAX uses only one port. I guess the down side to this would be that MY couldn't contact OTHER SIDE if OTHER SIDE dropped off, because it isn't working on standard port 5649. Correct? ___ --Bandwidth 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] random one way audio and noise between SIP phones on same LAN
Hi Klaus, I tried to set canreinvite=no following wiki advices but nothing changed. But I do not have a nat because I'm on a LAN. Is there any kind of program to control rtp packets flow? I think SIP conversation is established but RTP packets do not pass. I've tried to open up all ports (from 1020 to 65535) without success. These random one way calls ALWAYS happen about 10 times a day on EVERY PBX I have (on different LANs) and customers complain. I'm still searching for a monitoring application to catch packets on remote machines. Hope to find a good one. TIA Giorgio Incantalupo Klaus Darilion wrote: Do you use canreinvite (sip.conf)? Change the setting (setting canreinvite=yes may cause nat problems) nad verify if the problem still appears. Using htis setting you can find out if the Audio problem occurs only when media is relayed via Asterisk (-the problem is caused by Asterisk) or in all cases (the problem is not caused by Asterisk) regards klaus Giorgio Incantalupo wrote: Hi, sometimes I have one way calls and noise between sip phones connected to the same LAN so no nat/firewall is involved. I tried with different sip phone models soft phones and the result is the same. I searched inside every log file but found nothing. I made different PBX with different hardware but the result is always the same. Is there anybody experiencing this terrible problem? Considering to monitor a remote PBX via ssh, which text-only application could I use? TIA Giorgio Incantalupo ___ --Bandwidth 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] random one way audio and noise between SIP phoneson same LAN
I'm having the same random problem. I have canreinvite=no on all extensions. I have qualify = yes on all non-NAT extensions. I do have several NAT extensions, but I've not had reports of problems from those. 95% of my extensions (all polycom 501/601) are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches. In all cases, the called party cannot hear the calling party. The calling party has the still ringing icon on their phone, but can hear the called party talking. I've got call monitoring turned on, and asterisk is recording both sides of the conversation. The problem occurs on SIP-SIP and Zap-SIP calls. I've tried enabling sip debug on a particular extension that seemed to be experiencing the problem more than others. However the problem did not occur when the debugging was on. Sip debug generates so much noise I've been hesitant to turn it on system-wide. Is there a way I can turn on sip debug and have all that logging go to a specific file (and not in the asterisk console)? Also, are there any other configuration/logging tricks I can try? Thank you, Scott Scecina -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: Wednesday, October 18, 2006 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] random one way audio and noise between SIP phoneson same LAN Do you use canreinvite (sip.conf)? Change the setting (setting canreinvite=yes may cause nat problems) nad verify if the problem still appears. Using htis setting you can find out if the Audio problem occurs only when media is relayed via Asterisk (-the problem is caused by Asterisk) or in all cases (the problem is not caused by Asterisk) regards klaus Giorgio Incantalupo wrote: Hi, sometimes I have one way calls and noise between sip phones connected to the same LAN so no nat/firewall is involved. I tried with different sip phone models soft phones and the result is the same. I searched inside every log file but found nothing. I made different PBX with different hardware but the result is always the same. Is there anybody experiencing this terrible problem? Considering to monitor a remote PBX via ssh, which text-only application could I use? TIA Giorgio Incantalupo ___ --Bandwidth 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] Why is this happening?
Matt wrote: In the case of you example the IAX2 registration came in from the source port on the far device of 1207. Connections don't just move between ports. I understand all this. However, here is my question. MY on 4569 OTHER SIDE 1027. Is both the incoming and outgoing traffic on OTHER SIDE going in and out of 1027? I understand IAX uses only one port. I guess the down side to this would be that MY couldn't contact OTHER SIDE if OTHER SIDE dropped off, because it isn't working on standard port 5649. Correct? No. Asterisk will respond to the port that the registration came from. ___ --Bandwidth 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] Stopping putgoing calls after working hours
Thanx Jacob,
I did notice the locking phones at night mails in the list, and I have
just finished making the solution and it is just what I wanted here is
my addition in the extension.conf (with trixbox I did it in extensions_trixbox.conf),
note I have used the Authenticate command also with the astdb just serach
voip-info for the command and you will understand the drill, for each phone
number I added the password needed using the database put command
from console :
;MAG Addition for phone locking
;to lock extensions
exten => *00,1,Wait(1)
exten => *00,2,Answer
exten => *00,3,Authenticate(/${CALLERID(num)}|d) ; the caller will
be prompted for password
exten => *00,4,Set(DB(LOCKPHONE/${CALLERID(num)})=1) ; if passed
the lock variable will be set to equal one
exten => *00,5,Hangup
;to unlock extensions
exten => *01,1,Wait(1)
exten => *01,2,Answer
exten => *01,3,Authenticate(/${CALLERID(name)}|d) ; same as above
exten => *01,4,Set(DB(LOCKPHONE/${CALLERID(num)})=0); reverse of the
above
exten => *01,5,Hangup
All that is left is to make a line that checks the variable in astdb
when the calls are going to trunks and ofcourse made the trunk responsible
for emergency calls without this feature so you can only call the police,
ambulance, power and fire departments and internal extensions but not the
costly outside calls
Thx
MAG
Benjamin Jacob wrote:
Mohamed A. Gombolaty wrote:
> Dear Rich,
>
> It seems that my question is very general I apologize for that, but
I
> am glad to see others like yourself pointing me in different
> directions, it seems all around the world we have problems with the
> cleaning folks.
>
> What I have in mind is to make the phone user lock his phone when
he
> is leaving with a special code and relock it back when he comes to
> work (and
>
u mean unlock it..
> as for emergency calls there are attendants who work at night who
will
> be able to make an emergency call whenever needed at the spot), now
> there is nothing that seems to be able to do that directly, I have
> played around with the gotoiftime and also the time based dial plan
> include sent in mails before that.
>
> But while working I thought of another approach why not create a
php
> web interface that each user logs in with a special username and
> password and gives him access to lock his phone, and what php does
is
> actually change the secret password to something else than the
> configured on the phone, this should make the phone unable to
> authenticate thus not being able to make a call, and unlocking it
> returns the password to it's right form, I have already found the
> tables that I need to play around so I will restart making the php.
I
> will update the list back with my final result.
>
>
> Do you guys think I could send a mail to the dev site to see if they
> can add this feature to asterisk.
>
Am writing a few dialplans that you could use. I havent testted it..
u
might have to refine it.. am writing all this at runtime :-)
To lock and unlock phones, you need not go to php and change passwords
etc. You can use DB operations.
To lock phones, users can call into one particular number, e.g. *01
[lockphone]
exten => *01,1,Set(DB(LOCKPHONE/${CALLERID(num)})=1})
To unlock phones, u set the DB custom variable LOCKPHONE to zero,
using
another number, say *02
[unlockphone]
exten => *02,1,Set(DB(LOCKPHONE/${CALLERID(num)})=0})
So, to avoid calls, you'll have to check the value of this custom
variable everytime. To avoid repeated checks even in the day time,
you
can put the following dialplan, only in contexts which are invoked
at
night(read the previous posts).
[night-context]
exten => 911,1,Dial(Zap/999) ;;;wotever syntax, I've
never
worked with ZAP, for 911 emergency calls even at night.
include => lockphone
include => unlockphone
include => othernumbers
[othernumbers]
exten => _[0-9].,1, Set(locked=DB(LOCKPHONE/${CALLERID(num)}))
exten => _[0-9].,2,GotoIf($[${locked}=0]?:5)
allow call only
if phone is unlocked
exten => _[0-9].,3, Dial(SIP/${EXTEN})
phone is
unlocked , so call away to glory
exten => _[0-9].,4, Hangup
exten => _[0-9].,5, Playback(hussh-sleep-now) ;;; cant
call
now, cuz phones locked
exten => _[0-9].,6, Hangup
Now you lock n unlock ur phones whenever u want.
cheerz
- Ben.
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--
Thx
MAG
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[asterisk-users] Polycom IP650
Does anyone have an actual delivery date on the new Polycom HD IP650s? Im getting sick of not having a backlit screen and thinking of upgrading. Cheers, Dean ___ --Bandwidth 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] VoicePulse Connect 4 Channel Limit?
voipstreet allows 20 concurrent calls. On 10/16/06, Nate Kapi [EMAIL PROTECTED] wrote: Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. ___ --Bandwidth 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: Is 1.2.12.1 production ready (Mauro Zanin)
not having that issue with ChanSpy here but I loaded up a 1.2.12.1 box last night with a TE110P and Asterisk Crashes after receviing a call and I was using the latest zap drivers. I put in the Sangoma card and no problem. Must have been some motherboard compatibility. On 10/18/06, Thomas Kenyon [EMAIL PROTECTED] wrote: Mauro Zanin wrote:Hi Everybody, as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but VoiceMail application stated that there were no entries in voicemail.conf, so it didn't work. Installed again 1.2.0 and voilà the VoiceMail app. was working again. I asked to the group, but it seems I'm the only one with this issue! In Italy we say: Chi lascia la via vecchia per la nuova,sa quel che perde e non sa quel che trova litterally: Who leaves the old way, does know what he loses and doesn't know what he finds. Ciao MauroI've not had this problem, but I can say that with 1.2.12.1 if I usechanspy, when the spying handset hangs up asterisk segfaults (kickingall connected calls off).Finding that out was embrassing. (was on production server). ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 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
Re: [asterisk-users] Polycom IP650
Dean,I don't think anyone does yet, Polycom is telling us December.In the past, they've been pretty good at keeping their word. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Oct 18, 2006, at 8:43 AM, Dean Collins wrote: Does anyone have an actual delivery date on the new Polycom HD IP650’s? I’m getting sick of not having a backlit screen and thinking of upgrading.Cheers, Dean ___--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] Findme problem
Greetings all,
I've been working on having Asterisk put a call
through to two different numbers, and give the call to the first one that
acknowledges by pressing the 1 key. I found an example on the wiki, but I
can't get it working.
When I answer the call I hear the message telling
me to press 1 to connect, and as soon as the message is done, the call is
connected. In other words, it is not waiting for me to press a
key.
I'm sure this is a forehead slapper, but I just
can't see it...can anyone help? Here's the relevant portion of the
dialplan, It executes the NoOp(Waiting) and then the macro seems to
immediately exit and the call is connected.
[default]exten =
_XX,1,Dial(SIP/provider/${EXTEN:4},40,M(screen))exten =
_XX,2,Hangup
[macro-screen]exten = s,1,Wait(1)exten
= s,2,Set(TIMEOUT(digit)=5)exten =
s,3,Set(TIMEOUT(response)=10)exten = s,4,Background(press-1)exten
= s,5,NoOp(Waiting)
exten = 1,1,NoOp(Caller accepted)
exten = i,1,NoOp(Invalid response)
exten =
i,2,Set(MACRO_RESULT=CONTINUE)
exten = t,1,NoOp(Timeout)exten =
t,2,Set(MACRO_RESULT=CONTINUE)
[find-eric]exten =
s,1,Playback(pls-wait-connect-call)exten =
s,n,Dial(LOCAL/6135551212LOCAL/6135551313,40,m)
(I have replaced the phone numbers with bogus
ones).
Thanks,
Eric___
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Re: [asterisk-users] considering purchasing a t1 card, any recommendations?
I 4th it.On 10/18/06, Matthew Thompson [EMAIL PROTECTED] wrote: On 17 Oct 2006, at 22:09, Richard wrote: I would have to second the Sangoma buy. Their tech support is second to none and more then helpful. I've never had any problems with their products that wasn't my own fault. Thirded - I've just done another install with a Sangoma A102 - the setup guides you through all the way and takes no more than 30 minutes (Including recompiling zaptel, which it does for you) [EMAIL PROTECTED] :o) --Matthew Thompson[EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Fax: 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
RE: [asterisk-users] CAPI channel not available but nobody is usingthe system
Armin, I am running 1.2.7.1 with an Eicon T1 board version 2 on Debian 2.4 I don't know the details on chan-capi / CAPI drivers. We did the install April of this year. How can I tell what I have? Thank you for your time. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Armin Schindler Sent: Wednesday, October 18, 2006 3:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CAPI channel not available but nobody is usingthe system On Tue, 17 Oct 2006, Tim Sharp wrote: I have 23 CAPI channels defined and normally multiple channels are in use during the day for outbound calling. The problem is that every 3 or 4 months one of the channels becomes unavailable and then no calls can come in or go out on any of these channels. CAPI INFO shows Contr1: 23 B channels total, 22 B channels free. To fix the problem I reboot the asterisk server. First, is there a better way to reset the channels than rebooting? It depends where the problem really has its origin. If just asterisk (chan-capi) has a wrong channel count, it would be enough to unload chan-capi. Maybe asterisk itself need to be restarted. But if the real problem comes from the CAPI/ISDN driver, you need to reload these drivers. Which version of asterisk/chan-capi do you use? What ISDN hardware / CAPI driver do you use? Second, is there a way to bypass the unavailable channel in the dialplan? No. Third, what is causing the problem and can I prevent it? chan-capi counts the active channels when the CONNECT/DISCONNECT message of b-channels are indicated. If one of these messages are missing (it's a bug in the CAPI driver if that happens) the count is wrong. Armin ___ --Bandwidth 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] Electric usage of a tdm400p
Well Im planning to use a mini-itx, a laptop hdd and a 4fxs digium card. the mini-itx comes with a 60W DC to DC adapter (80W peak). So I need power to manage the hdd, motherboard,the tdm card. A disk cable can be made available, but is not present as a factory default. So My real concern is power. On 10/18/06, Bob Chiodini [EMAIL PROTECTED] wrote: On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote: Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? Erick, Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN ~5). This translates to 2.7 watts. Assuming a DC/DC converter efficiency of 38% (probably low), you would need about 3.7 watts, per FXS module. About 15 watts, total. What is the TDM card installed in and is a disk drive cable available? Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth 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] QueueMetrics 1.3 released today
Hello list, I am pleased to tell you that we have released a new version of QueueMetrics. The main areas of improvement were the following ones: - Internationalization engine: the text of QueueMetrics can be easily localized, as well as numbers and dates. - Internationalized versions: Italian, German and French - Export all data to Excel, CSV files or XML - Database update utility: makes the database up-to-date to the latest version of QM. - You can now connect to an Asterisk server for call barge-in by using the Manager interface as well as call-files. - You can now create a single-URL login and show realtime screens for kiosks or wallboards. A full list of improvements over version 1.2.1 can be found at http://queuemetrics.loway.it/news.jsp QueueMetrics 1.3 allows data storage on both flat files and MySQL databases for bigger call centers. And of course comes with a 90-page user manual that covers all aspects of it. QueueMetrics is a commercial call center monitoring package, but is availabe free of charge for individuals, Asterisk hackers and small SOHOs. You can request a trial key if you run a larger installation and would like to test it in your own environment. The latest version of QueueMetrics can be downloaded from http://queuemetrics.loway.it/download.jsp Hope you like it, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth 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] Is 1.2.12.1 production ready
No need for a religious argument! But my OS has been ordained by GOD! j/k I have, on occasion, had to reboot Linux CentOS 3.x with Asterisk 1.2.10, but certainly not daily, once a week or longer. Only when something is obviously insane. I run Tao Linux on a number of the asterisk boxes I administer, and it's basically the same thing as CentOS. To get it to the point of useable stability, I had to disable probably about 20 or so useless processes that are running by default. Before I disabled these processes, I did have to schedule weekly reboots. I'm now able to keep it running continuously running without hiccups with all the versions of asterisk that I mentioned previously in this thread. The only reboots I have are for occasional security updates, about once every few months or so. I also did run CVS-HEAD (pre 1.2) in production at one point, and that WAS pretty hairy as far as stability is concerned. When I did that, I did have to run cron scripts to restart asterisk, but not the whole system. I definitely agree with other posters in this thread that reboots are generally NOT the answer, at least as a long term solution. Just from anecdotal experience, it seems more likely that you'll run into problems at boot time than if you leave the system running and just restart certain offending processes. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] list down?
Im not getting any asterisk emails this morning even though they are being delivered to the archives http://lists.digium.com/pipermail/asterisk-users/2006-October/thread.html Cheers, Dean ___ --Bandwidth 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] random one way audio and noise between SIP phones on same LAN
On Wed, Oct 18, 2006 at 01:44:06PM +0200, Giorgio Incantalupo wrote: sometimes I have one way calls and noise between sip phones connected to the same LAN so no nat/firewall is involved. I tried with different sip phone models soft phones and the result is the same. I searched inside every log file but found nothing. Plug everything into a hub, not a switch, and then run tcpdump to look at the traffic going back and forth: # tcpdump -i eth0 -n -s0 -v Look at the RTP packets, in particular source and destination IP addresses. Do they go phone1-phone2 and phone2-phone1 ? Or do they go phone1-PBX, PBX-phone2, phone2-PBX and PBX-Phone1 ? In the latter case, it's not the phones which are at fault, it's your PBX. If the PBX is Asterisk, and you want help solving the problem, then you'd better give full details of your system (hardware and O/S platform, asterisk version, how it's configured) If you want the streams to go direct between phones, and the PBX is Asterisk, then set canreinvite=yes in sip.conf Considering to monitor a remote PBX via ssh, which text-only application could I use? Depends what the PBX is. If it's something with a serial console, then minicom, tip or cu will let you talk to it. If it's asterisk, then use the asterisk console (asterisk -r) HTH, Brian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX Terminal
Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil ___ --Bandwidth 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] random one way audio and noise between SIP phoneson same LAN
Hi Scott, seems that we have the same problem...I have canreinvite=no and polycom phones. I do not have cisco switch and qualify=yes but I think that is not the problem. I've got 2 questions: 1) my polycom firmware is: sip.ver: 1.6.5.0043 bootrom.ver: 2_6_2 what are yours? 2) have you got one way calls only or noise on sip calls conversations too? TIA Giorgio Incantalupo P.S.: for configuration/monitoring apps I'm still on it...I hope to find useful tools asap. In case, I'll let you know. Scott Scecina wrote: I'm having the same random problem. I have canreinvite=no on all extensions. I have qualify = yes on all non-NAT extensions. I do have several NAT extensions, but I've not had reports of problems from those. 95% of my extensions (all polycom 501/601) are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches. In all cases, the called party cannot hear the calling party. The calling party has the still ringing icon on their phone, but can hear the called party talking. I've got call monitoring turned on, and asterisk is recording both sides of the conversation. The problem occurs on SIP-SIP and Zap-SIP calls. I've tried enabling sip debug on a particular extension that seemed to be experiencing the problem more than others. However the problem did not occur when the debugging was on. Sip debug generates so much noise I've been hesitant to turn it on system-wide. Is there a way I can turn on sip debug and have all that logging go to a specific file (and not in the asterisk console)? Also, are there any other configuration/logging tricks I can try? Thank you, Scott Scecina -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: Wednesday, October 18, 2006 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] random one way audio and noise between SIP phoneson same LAN Do you use canreinvite (sip.conf)? Change the setting (setting canreinvite=yes may cause nat problems) nad verify if the problem still appears. Using htis setting you can find out if the Audio problem occurs only when media is relayed via Asterisk (-the problem is caused by Asterisk) or in all cases (the problem is not caused by Asterisk) regards klaus Giorgio Incantalupo wrote: Hi, sometimes I have one way calls and noise between sip phones connected to the same LAN so no nat/firewall is involved. I tried with different sip phone models soft phones and the result is the same. I searched inside every log file but found nothing. I made different PBX with different hardware but the result is always the same. Is there anybody experiencing this terrible problem? Considering to monitor a remote PBX via ssh, which text-only application could I use? TIA Giorgio Incantalupo ___ --Bandwidth 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
[asterisk-users] DTMF problems with legacy PBX
I am having trouble with a Panasonic D500 that is connected to an Asterisk server via an E1 and then the Asterisk server dials out via a SIP provider. The problem is that DTMF is not being recognized when they try to call an external IVR like a bank service. If I connect a SIP phone to the Asterisk server and dial using the SIP line I can use the external IVR, but when al call is placed from a phone connected to the Panasonic D500 the IVR will not respond. I am using Asterisk 1.2.10 on CentOS 4.4 and the Panasonic is connected using Unicall on a TE110P. I know DTMF is fine from the Panasonic to the Asterisk because I can dial the Voicemail extension and navigate the menu, it is just when we try to get an external IVR that we have a problem. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth 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+SER help
On Wed, Oct 18, 2006 at 05:31:52AM -0700, Crazy Boy wrote: I want to setup multiple SIP accounts. How can I do this? That depends what you mean by setup multiple SIP accounts. I'm not a mind reader, but I can think of two possibilities: (1) You want to have multiple phones on the Asterisk server (registering to Asterisk as separate SIP accounts), and a single sipgate.co.uk link to the outside world. Incoming calls to your sipgate number should ring all phones, or ring the phones in sequence. This is just a question of setting up the dialplan correctly. sipgate.co.uk is a fine service, and in particular you can place multiple outbound calls concurrently and have multiple inbound calls to the same phone number (DDI). In fact, you don't need Asterisk at all; you can just configure all your phones to register directly with sipgate, which does call forking (i.e. calls to your DDI will ring all registered phones simultaneously). But then you are limited to sipgate's voicemail etc. (2) You want to have multiple sipgate.co.uk accounts - so sipgate gives you multiple DDIs - and have Asterisk register itself multiple times to sipgate. That's just setting up the accounts in sip.conf and suitable dialplan config, so that for example calls to DDI 1 ring one set of phones and calls to DDI 2 ring another set of phones (or whatever it is you're trying to accomplish by having multiple sipgate accounts) I have installed Asterisk, created Asterisk SIP extensions and registered in www.sipgate.co.uk. Now, what I have to do? I don't know. What does it do now, and what would you like it to do differently? 1) Am I need to install SER or OpenSER in my server along with Asterisk? Almost certainly not needed. Pretty much any scenario you can think of, Asterisk can do. SER/OpenSER can be used if you are doing a large volume of SIP-to-SIP routing (hundreds or thousands of call setups per second) 3) I searched in Internet. But, I didn't find good tutorial for this. www.asteriskdocs.org, chapters 5 and 6 are all about dial plans. www.voip-info.org for setting up chan_sip Brian. ___ --Bandwidth 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] Why is this happening?
On Wed, Oct 18, 2006 at 09:11:15AM -0400, Matt wrote: In the case of you example the IAX2 registration came in from the source port on the far device of 1207. Connections don't just move between ports. I understand all this. However, here is my question. MY on 4569 OTHER SIDE 1027. Is both the incoming and outgoing traffic on OTHER SIDE going in and out of 1027? Packets from the other side to you will have source IP x.x.x.x source port 1027 destination IPy.y.y.y destination port 4569 Packets from you to the other side will have source IP y.y.y.y source port 4569 destination IPx.x.x.x destination port 1027 If there is NAT in between, then the packets may have their source and/or destination address and/or port changed by the time they reach the other side. This depends on how the NAT is set up, and which device is on the inside of the NAT and which is on the outside. ___ --Bandwidth 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] Netgear WGT Flash-fest at Astricon
Brian Capouch wrote: Just an FYI to anyone out there who will be attending Astricon and who would like to play around with embedded Asterisk on the Netgear WGT634U platform. If you want to bring your own to the show, I'll be bringing all the appropriate stuff to flash them there with my latest openWGT/Asterisk build. They are available from www.justdeals.com, refurbs, for $44.95 delivered. You'll also need a USB flash drive. I use 256MB, but Asterisk can be set up to use as little as 32MB. B. B, Do you have the necessary components for a serial cable for these little guys? I would like to play with the loader and get a serial console... If you don't have one perhaps we can work on getting the parts before then. -- 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] IAX Terminal
Lets change the question to : does somebody know good iax phones, that are ROHS compliant and without enormous delivery problems ? Neil Tancock wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil ___ --Bandwidth 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 config chanspy
How can I do to select the channel to spy ? thanks De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Ralph Liebessohn Enviado el: Miércoles, 18 de Octubre de 2006 09:29 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] how to config chanspy On 10/17/06, Thirumal Saminathan [EMAIL PROTECTED] wrote: hi all, please any one help me ,how to configure chanspy application . and also send me if u have any sample configure file. -thiru Hi, It could be very simple, like: exten = 123,1,ChanSpy() ; Spy all channels or more accuracy: exten =124,1,ChanSpy(SIP) ; Spy all sip channels if I can help you more, let me know! -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --Bandwidth 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] ooh323 dtmf problem
anybody successfully running asterisk-callmanager scenario with h323 trunk (ooh323 channel driver in asterisk)? I'm using 1.2.12.1 ooh323 from 1.2.4 add-ons, but seems, that ooh323 is ignoring dtmf digits from callmanager h323 trunk setup with chan_h323 is working fine with dtmf I tried all possible modes with ooh323, but without success, with chan_h323, I'm using default (rfc2833) and it works possible dtmf modes from chan_ooh323 ; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad ; h245alphanumeric, h245signal. ;Default - rfc 2833 dtmfmode=rfc2833 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium on Dell PowerEdge 1850
We're running 2 TE412P's in a Dell 1850 just fine, been running like this for well around 6 months to a year now without any problems. They're not exactly 212P's but I imagine it won't be much different. On Wed, 2006-10-18 at 10:54 +0200, Tomislav Parčina wrote: Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Best regards, -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gotoiftime and Macro question
You don't run a function in the GotoIfTime application, you point to another context/extension/priority to jump to that DOES have the applications you need, as Conrad exampled. Moj [EMAIL PROTECTED] wrote: Is there a way to run a macro in a GotoIfTime statement ?? from the wiki documentation it seems not, but.. I would like to do something like this: . 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?Macro(exten-vm,novm,567)) it does not work, as expected from documentation any workaround to call an extension WITHOUT vm (also if vm for that extension is present...) as a consequence of a Time condition? thanks in advance Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:500,45361734218322068143078! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth 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: ZapHFC quadBRI D-Channel going down randomly
I have the exact same problem on a normal ISDN2 BRI line. I solved it by having my Telco put layer 1 to permanent. Best regards, Henrik Woffinden Alberto Pastore wrote: asterisk ha scritto: On most traditional pabx's it's possible to set layer 1 to permanent or call. It sounds like your system is configured for permanent and your lines to call. How you would set this on asterisk I have no idea. fadge The question is: is it possible I am the only one with such problems on all asterisk boxes on different sites and different ISDN lines? I've googled around on many forums but no one seems to have this one. The old replaced PBXs had layer 1 set for call, as you say, and they showed no problems at all. With asterisk as a PBX, every 2-3 hours, you cannot dial out for 5 to 15 minutes then everything gets back to normal (no idea about what triggers the return to working state). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Pastore Sent: 16 October 2006 17:26 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ZapHFC quadBRI D-Channel going down randomly Hi. I'm running some asterisk boxes on different sites, some equipped with a couple of ZapHFC cards, others with Junghanns quadBRI cards. All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6) and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with kernel 2.6.17.3 The cards are connected to Telecom Italia's NT1/NT1+ S/T lines; some of them are point-to-point, others are point-to-multipoint. I keep getting always the same problem: after some hours of regular working, some boxes report the usual message Primary D-Channel on span n down (where n is different every time, depending on the number of active bri spans) I've read on previous postings that having layer 1 down on ptmp spans is normal. However after getting a down message (on ptp spans too!) I'm no more able to place outgoing calls on that span, until I restart asterisk zaptel drivers. Sometimes, they get back working by themselves (with the related span up notification) after a random time period. During the down period, incoming calls are regularly served. However these calls do not change the status of the span, i.e. as soon as the calls are hung up, the span gets down again. I've tried to capture the dialog between the card and NT1 equipment, and during the down state, I got this repeated over and over: Sending Set Asynchronous Balanced Mode Extended [ 00 8b 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 069EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] == Primary D-Channel on span 1 down In zapata.conf I'm pretty sure I've always set the correct signalling settings (switchtype = euroisdn, signalling = bri_cpe_ptmp or bri_cpe depending on the case) In /etc/zaptel.conf, I've tried many combinations with no difference; my current settings are like this: span=1,1,0,ccs,ami bchan=1-2 dchan=3 span=2,1,0,ccs,ami bchan=4-5 dchan=6 etc Any clue? Thanks, Alberto -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth 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] Electric usage of a tdm400p
I set up a similar system on an VIA Epia 5000, and I had issues when I included the CDROM in the mix. I had to use another ATX power supply to complete the install, but then once I removed the CDROM drive I had no power issues. I presume you could install the OS with the CDROM drive installed and the molex power connector REMOVED from the TDM card, then when the OS was installed and you had network connectivity, power down, remove the CDROM, add the power supply for the TDM card, then install zaptel etc. Or just try it and tell us what happens, low power won't break it in my experience. Your cdrom drive might have a lower power consumption than mine. Moj Erick Perez wrote: Well Im planning to use a mini-itx, a laptop hdd and a 4fxs digium card. the mini-itx comes with a 60W DC to DC adapter (80W peak). So I need power to manage the hdd, motherboard,the tdm card. A disk cable can be made available, but is not present as a factory default. So My real concern is power. On 10/18/06, Bob Chiodini [EMAIL PROTECTED] wrote: On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote: Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? Erick, Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN ~5). This translates to 2.7 watts. Assuming a DC/DC converter efficiency of 38% (probably low), you would need about 3.7 watts, per FXS module. About 15 watts, total. What is the TDM card installed in and is a disk drive cable available? Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth 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 with fxotune
What does fxotune need to do its job correctly - complete silence on the line, or just absence of the dial tone? For how long? I've been trying to get a silent termination number out of my telco all morning and have come up empty. When I just dial 1 as some online guides suggest to break the dial tone, it is silent for a few seconds, then plays a Your call cannot be connect... message, which I assume would interfere with fxotune. Likewise for using 4, 5', etc as I have seen suggested elsewhere. -j -- Jeremy Jongsma [EMAIL PROTECTED] Traders Media ___ --Bandwidth 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 config chanspy
seriously go look at voip-info.org for the answer, thats where we get most of our info from, or perhaps type show application chanspy from the asterisk CLI.Are we that lazy that we cant use google to search. Ridiculous. On 10/18/06, Sergio R. D'Ippolito [EMAIL PROTECTED] wrote: How can I do to select the channel to spy ? thanks De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] En nombre de Ralph Liebessohn Enviado el: Miércoles, 18 de Octubre de 2006 09:29 a.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] how to config chanspy On 10/17/06, Thirumal Saminathan [EMAIL PROTECTED] wrote: hi all, please any one help me ,how to configure chanspy application . and also send me if u have any sample configure file. -thiru Hi, It could be very simple, like: exten = 123,1,ChanSpy() ; Spy all channels or more accuracy: exten =124,1,ChanSpy(SIP) ; Spy all sip channels if I can help you more, let me know! -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___--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] Locking phones at night...
Tzafrir Cohen wrote: Is there a simple and safe way to query the astdb database outside of Asterisk? after writing to it with: asterisk -rx 'database put phones locked 1' something like asterisk -rx 'database get phones locked' returns 1... Is this what you mean by outside of asterisk? Sorry if I misunderstood. I'm under the impression (possibly erroneously) that asterisk doesn't flush its database to disk often enough for you to trust copies that might be stored there, or to notice new changes made on-disk by you. That being said, Berkeley DB file, /var/lib/asterisk/astdb. Moj -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX softphones
Message: 16 Date: Wed, 18 Oct 2006 16:10:38 +0100 From: Neil Tancock [EMAIL PROTECTED] Subject: [asterisk-users] IAX Terminal To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil Neil, www.asteriskguru.com lists a few of them. Try IDEFISK. Paul Gaffney LANStatus,LLC ___ --Bandwidth 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] random one way audio and noise between SIP phoneson same LAN
Scott Scecina wrote: In all cases, the called party cannot hear the calling party. do you have the RTP ports open? signature.asc Description: OpenPGP digital 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] Why is this happening?
Brian Candler wrote: On Wed, Oct 18, 2006 at 09:11:15AM -0400, Matt wrote: In the case of you example the IAX2 registration came in from the source port on the far device of 1207. Connections don't just move between ports. I understand all this. However, here is my question. MY on 4569 OTHER SIDE 1027. Is both the incoming and outgoing traffic on OTHER SIDE going in and out of 1027? Packets from the other side to you will have source IP x.x.x.x source port 1027 destination IPy.y.y.y destination port 4569 Packets from you to the other side will have source IP y.y.y.y source port 4569 destination IPx.x.x.x destination port 1027 If there is NAT in between, then the packets may have their source and/or destination address and/or port changed by the time they reach the other side. This depends on how the NAT is set up, and which device is on the inside of the NAT and which is on the outside. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users it's kind of like taking a flight.. you enter the airport (make the call) through the main entrance (port 4569). then to get to your flight (establishing the call) you have to go to the terminal (port 1027) to get to your destination (the called number) signature.asc Description: OpenPGP digital 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] random one way audio and noise betweenSIP phoneson same LAN
Giorgio, I'll answer in reverse order: I've not had reports of noise from my users. However, when I went down to get the s/w version from the phone that seems to be acting up the most, the user reported that earlier they were actually on a call that was ok then spontaneously dropped the audio. Per my instructions (based on another similar report I read on Digium's site), my user hit a digit on the phone which brought back the caller's audio. I've also had them attempt to put the call on hold, and then resume, but that did not bring the audio back. As far as the S/W versions: One of the phones that acts up (and they all should match): Polycom 501 BootRom: 3.1.3.0131 BootBlock: 2.5.0 SIP: 1.6.6.0036 My phone, on which I've never experienced the problem: Polycom 601 BootRom: 3.1.3.0131 BootBlock: 2.6.0 SIP: 1.6.6.0036 - Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Wednesday, October 18, 2006 11:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] random one way audio and noise betweenSIP phoneson same LAN Hi Scott, seems that we have the same problem...I have canreinvite=no and polycom phones. I do not have cisco switch and qualify=yes but I think that is not the problem. I've got 2 questions: 1) my polycom firmware is: sip.ver: 1.6.5.0043 bootrom.ver: 2_6_2 what are yours? 2) have you got one way calls only or noise on sip calls conversations too? TIA Giorgio Incantalupo P.S.: for configuration/monitoring apps I'm still on it...I hope to find useful tools asap. In case, I'll let you know. Scott Scecina wrote: I'm having the same random problem. I have canreinvite=no on all extensions. I have qualify = yes on all non-NAT extensions. I do have several NAT extensions, but I've not had reports of problems from those. 95% of my extensions (all polycom 501/601) are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches. In all cases, the called party cannot hear the calling party. The calling party has the still ringing icon on their phone, but can hear the called party talking. I've got call monitoring turned on, and asterisk is recording both sides of the conversation. The problem occurs on SIP-SIP and Zap-SIP calls. I've tried enabling sip debug on a particular extension that seemed to be experiencing the problem more than others. However the problem did not occur when the debugging was on. Sip debug generates so much noise I've been hesitant to turn it on system-wide. Is there a way I can turn on sip debug and have all that logging go to a specific file (and not in the asterisk console)? Also, are there any other configuration/logging tricks I can try? Thank you, Scott Scecina ___ --Bandwidth 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] Unique ID
On 17 Oct 2006, at 19:49, Eric Rousse wrote: Hello guys, We're currently working on asterisk trying to create our own SIP phone, because we need special features. But dunno maybe there's other people who already done that before. Basically, we are a inbound call center. We have serveral customers with different phone numbers, which are redirected to us. When we receive a call coming on a specific phone number, the call gets identified with the number and there's a greeting associated and displayed on the agent soft phone(this technology is still using regular phone with a special computer device). But here's the challenge that we currently face: 1. We need to have the info for the hold time (from agent) and hold time(before the call is actually answered). We currently offer a different pricing for the hold time by an agent than from the other one hold time. 2. We're currently trying to identify calls by unique id for billing, I've found about that the variable $UNIQUEID which I could use, and there's also the cdr table that I can create, but it would be nice to have both in the cdr table ? That way I could probably create a second table in the asterisk db, and store our hold time, sent from the softphone. Anyway, does all that ring a bell to someone ? Something that was already done ? It rings some bells, we have done something similar (but not identical). You can configure custom cdr to include the unique ID. You could set the callerName field to the uniqueID and your softphone will then be able to post to the second table using the uniqueID as a cross reference . You can't make uniqueID a foreign key in the second table as asterisk does not write CDRs synchronously - meaning that the cdr may not be in the database (yet) when your softphone posts it's data (you will probably need to put an index on it to get decent performance) When it comes time to implement your softphone, please give our Corraleta Technology SDK a look it is designed for this sort of thing - fully customizable and scriptable. Tim. Tim Panton www.mexuar.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] install MAGI
On 17 Oct 2006, at 17:14, Alvaro A Colunga Rdz wrote: Hi, can somebody point me where to get MAGI patch to run AGI commands through asterisk manager. What i need to do is play a sound after originating a call on a zap channel. Or if its another way for doing this can somebody tell me. It depends on exactly what you want.. You may be able to do this with the Local channel and manager Oringinate. Tim. Tim Panton www.mexuar.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] random one way audio and noise betweenSIP phoneson same LAN
Yes, 1-2 are open. This phenomenon is random. Most calls work fine most of the time. - Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten Sent: Wednesday, October 18, 2006 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] random one way audio and noise betweenSIP phoneson same LAN Scott Scecina wrote: In all cases, the called party cannot hear the calling party. do you have the RTP ports open? ___ --Bandwidth 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] IAX softphones
On Wed, October 18, 2006 19:03, Paul Gaffney wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil Neil, www.asteriskguru.com http://www.asteriskguru.com/ lists a few of them. Try IDEFISK. Paul Gaffney LANStatus,LLC I personally like DIAX on for Windows users. Haven't yet found an IAX phone I like on Linux... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth 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] IAX softphones
On Wed, 2006-10-18 at 20:08 +0200, Francesco Peeters (Asterisk) wrote: On Wed, October 18, 2006 19:03, Paul Gaffney wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil Neil, www.asteriskguru.com http://www.asteriskguru.com/ lists a few of them. Try IDEFISK. Paul Gaffney LANStatus,LLC I personally like DIAX on for Windows users. Haven't yet found an IAX phone I like on Linux... Kiax works great with Gnome, KDE or Xfce. -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Card
Hi - We are looking at migrating our office from a Samsung PBX to an Asterisk PBX. I am looking at ordering a PRI with 12 Channels for now (we currently have 8 analog lines) and need to know what PRI card you guys would recommend that we use. I have seen some with Echo Cancellation and so on, but don't know which one would be best to get. Your main choices for an internal PRI card are Digium and Sangoma, though Rhino is now offering one, too. Digium, Rhino and Sangoma all offer 1, 2, and 4 port versions. Digium and Sangoma offer them with or without hardware echo cancellation. Sangoma also offers an 8 port version (with or without echo can). I offer this information in an unbiased way in the hopes of preventing a holy war. I can't speak for the Rhino cards as they are rather new, but both the Digium and Sangoma cards work well. Many Sangoma users feel that their cards are compatible with a wider variety of motherboards, but that's the subject of much debate on this list. I administer a couple of systems with Digium TE410p's, and they work well. Digium offers unlimited support up until their cards are installed and working. They'll RMA most any non-functioning card. Sangoma offers a 5 year warranty, and will generally help you out whenever you need it. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Go to DIGIUMBOARDS.COM
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 [EMAIL PROTECTED] wrote: Best prices World-Wide AGAIN?! I thought they were banned?! - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFNn85S6d5vy0jeVcRAk/TAJ9NgO83AW401U2tLi9SCI6hhs/SHwCfRlgR xo5vGR3teDcfJuPUkYxXkWI= =MR7y -END 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
[asterisk-users] Sip Trunks
Hello, well, I need to configure two asterisk box like SIP trunks to send sip calls from one asterisk to the other and visceversa. So How I setup config files to get this working?.Thanks. G. ___ --Bandwidth 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] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M
On Wed, 18 Oct 2006, Klaus Darilion wrote: Hi (Armin)! Does someone knows how to identify the type of the card? The delivery note says it is a V-4BRI-8M, whereas lspci reports a 4BRI-8M. What is it really? Are there any Eicon tools to identify the card type? As far as I know these cards are almost identical, but the PCI ID must be different. Maybe the pci id database doesn't have this difference... What PCI-ID does it have? 0xE012 = 4BRI-8M 0xE013 = 4BRI-8M V2 0xE016 = Voice 4BRI-8M 0xE017 = Voice 4BRI-8M V2 There is no special tool. When you load the divas driver, it should announce the cards found. And the divactrl utility uses divas to get the cards info and can tell the correct version as well, e.g.: divactrl ctrl -c 1 -CardName Armin ___ --Bandwidth 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] CAPI channel not available but nobody is usingthe system
On Wed, 18 Oct 2006, Tim Sharp wrote: Armin, I am running 1.2.7.1 with an Eicon T1 board version 2 on Debian 2.4 I don't know the details on chan-capi / CAPI drivers. We did the install April of this year. How can I tell what I have? The divas driver version can be found in the syslog messages when the driver is loaded. I recommend to use the new V3 driver (ftp.melware.net). When you start asterisk (with verbosity 5) you can see the chan-capi messages including its version. It's an too old version if it is from April, please update, same ftp-server. Armin Thank you for your time. Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Armin Schindler Sent: Wednesday, October 18, 2006 3:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CAPI channel not available but nobody is usingthe system On Tue, 17 Oct 2006, Tim Sharp wrote: I have 23 CAPI channels defined and normally multiple channels are in use during the day for outbound calling. The problem is that every 3 or 4 months one of the channels becomes unavailable and then no calls can come in or go out on any of these channels. CAPI INFO shows Contr1: 23 B channels total, 22 B channels free. To fix the problem I reboot the asterisk server. First, is there a better way to reset the channels than rebooting? It depends where the problem really has its origin. If just asterisk (chan-capi) has a wrong channel count, it would be enough to unload chan-capi. Maybe asterisk itself need to be restarted. But if the real problem comes from the CAPI/ISDN driver, you need to reload these drivers. Which version of asterisk/chan-capi do you use? What ISDN hardware / CAPI driver do you use? Second, is there a way to bypass the unavailable channel in the dialplan? No. Third, what is causing the problem and can I prevent it? chan-capi counts the active channels when the CONNECT/DISCONNECT message of b-channels are indicated. If one of these messages are missing (it's a bug in the CAPI driver if that happens) the count is wrong. Armin ___ --Bandwidth 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] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M
On Wednesday 18 October 2006 04:52, Klaus Darilion wrote: Hi (Armin)! Does someone knows how to identify the type of the card? The delivery note says it is a V-4BRI-8M, whereas lspci reports a 4BRI-8M. What is it really? Are there any Eicon tools to identify the card type? thanks klaus :0a:03.0 Network controller: Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 (rev 01) Subsystem: Eico Have you tried running the update-pciids command to make sure your pci-id database is current? It's part of the pciutils package. -- ~~~ Carla Schroder check out my Linux Cookbook, the ultimate Linux user's and sysadmin's guide! http://www.oreilly.com/catalog/linuxckbk/ ~~~ ___ --Bandwidth 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] Why is this happening?
On Tue, Oct 17, 2006 at 11:20:02AM -0500, Mitch Miller wrote: Sockets and Ports often gets confused with each other. I feel like I'm reading Suess... Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth 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] 1.4 downgrade
I am having a bunch of issues with 1.4 and want to go back to 1.2 any ideas on the best way I saw someone say apt-get remove will this work for asterisk or do I need to do it for each libpri, addons, zaptel and asterisk? Thanks Jason ___ --Bandwidth 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] adding outbound prefix
Does anyone know how I can add a prefix to an outbound SIP call? I believe this would be done in extensions.conf, but am not sure how to go about it. Thanks ___ --Bandwidth 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] IAX softphones
On Wed, October 18, 2006 21:07, Guillermo Salas M. wrote: On Wed, 2006-10-18 at 20:08 +0200, Francesco Peeters (Asterisk) wrote: On Wed, October 18, 2006 19:03, Paul Gaffney wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil Neil, www.asteriskguru.com http://www.asteriskguru.com/ lists a few of them. Try IDEFISK. Paul Gaffney LANStatus,LLC I personally like DIAX on for Windows users. Haven't yet found an IAX phone I like on Linux... Kiax works great with Gnome, KDE or Xfce. -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 I'll try that later, thanks! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth 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] echotraining=yes in misdn.conf is invalid or out of range.
Hi. I'm having problems with chan_mISDN configuration. Line echotraining=yes causes warning, when Asterisk is parsing misdn.conf and I'm confused why the PBX doesn't accept the setting. No matter which section I try to offer it, it is always invalid or out of range. The setting itself is supposed to be valid, it is in the sample configuration file of chan_mISDN 0.3.1. When I list the configuration of the ISDN-ports with misdn show config, I can find values for echocancel and echocacelwhenbridged, but no mention about echotraining. I'm running Asterisk 1.2.10 on OpenSuse 10.1 with chan_misdn-0.3.1-rc23. The hardware platform is HP server with Intel XEON processor and three hfcpci BRI -cards in TE -mode. - Here's the message I get: == Parsing '/etc/asterisk/misdn.conf': Found Oct 19 00:33:54 WARNING[4443]: misdn_config.c:660 _build_port_config: misdn.conf: echotraining=yes (section: default) invalid or out of range. Please edit your misdn.conf and then do a misdn reload. - And here is my misdn.conf: [general] debug=1 bridging=no tracefile=/var/log/asterisk/misdn.trace [default] echocancel=yes echotraining=yes hold_allowed=yes screen=-1 presentation=-1 senddtmf=yes [isdn_call] ports=1,2,3 context=isdn_in msns=* - Have you got any idea what is causing this and how I could get the echo training working? Thank you for your help. Jarkko ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4 downgrade
On Wednesday 18 October 2006 14:31, Jason Walker wrote: I am having a bunch of issues with 1.4 and want to go back to 1.2 any ideas on the best way I saw someone say apt-get remove will this work for asterisk or do I need to do it for each libpri, addons, zaptel and asterisk? That works only if you are running Debian or some flavor thereof like Ubuntu, and installed it with apt-get. Otherwise it's just source installs. 1.4 has a make-uninstall target: [EMAIL PROTECTED] asterisk-1.4.0-beta2]# make uninststall ... [stuff and more stuff] ... +- Asterisk Uninstall Complete -+ + Asterisk binaries, sounds, man pages, + + headers, modules, and firmware builds,+ + have all been uninstalled.+ + + + To remove ALL traces of Asterisk, + + including configuration, spool+ + directories, and logs, run the following + + command: + + + +make uninstall-all + +---+ I haven't looked in the other source directories, but it's easy enough to try out. You can always read the Makefiles to see what they can do. -- ~~~ Carla Schroder check out my Linux Cookbook, the ultimate Linux user's and sysadmin's guide! http://www.oreilly.com/catalog/linuxckbk/ ~~~ ___ --Bandwidth 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] Windows and file shares
I think this went to the wrong list. This is asterisk support, not samba. Having said that, I'd be happy to take a look if you want me to ssh in. I have 6 samba boxes in as many states. Ejay Hire [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul GaffneySent: Wednesday, October 18, 2006 7:36 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Windows and file shares Message: 12 Date: Tue, 17 Oct 2006 18:07:04 -0700 (PDT) From: sdgesa gaeharth [EMAIL PROTECTED] Subject: Re: [asterisk-users] Extremely choppy sound on some of ourPOTSnetwork calls; goes away with mute To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset="iso-8859-1" None of these steps have made a difference. Any other suggestions? Here is my original post: Can anyone help me to figure out why I can not write to a public share? I was able to join the domain without a problem. I can access the share from an xp box. I just can not write: "Access denied". thanks On Windows check two sets of permissions: The share permission and the file permission. By default on a Windows 2003 server the share permission is set to read only. You need to change it to read/write. Even if you have the file permission set to full control for everyone - the share permission of read only will block a user from changing a file. Windows 2000 or older did not default to read only on the share but it would be worth looking at if that is the OS you are using. Paul Gaffney LANStatus, LLC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
