Re: [asterisk-users] MYSQL problem
2010/1/27 Steve Edwards asterisk@sedwards.com: Un-mid-posting... On Fri, 22 Jan 2010, Zhang Shukun wrote: as you know, we can use MYSQL command to visit mysql database but if i use other database like Oracke,sybase,etc, Could i use MYSQL command ? 2010/1/23 Steve Edwards asterisk@sedwards.com: ODBC will do what you want. Personally, I'd vote for an AGI using whatever C API your DB provides -- like Pro*C to access Oracle. You will have access to all of the features of your DB and your dialplan will be a lot cleaner and easier to maintain. On Wed, 27 Jan 2010, Zhang Shukun wrote: Thanks, while i think because oracle has no offical ODBC for linux system. is that better use mysql than oracle. considering the perfoermace and speed. many people around think mysql is not a good option for database, they think mysql is only suit for small business. but i want to have a try. i need to convince them to use this. So don't use ODBC, use Pro*C... you said Personally, I'd vote for an AGI using whatever C API your DB provides what do you think about phpagi and cagi, if i choose the agi method. while phpagi seems used more popular than cagi. Back when Yahoo was relevant, they ran on MySQL. Can you quantify your requirements (number of rows, queries per second, simultaneous connections) and test it on hardware similar to your production environment? i cant quantify my requirement now. but the business has not be start. and the user will increase as time going. While I'm sure Yahoo spent a lot of time and money designing and tuning their system, sometimes explain plan can point you to small changes that yield significant results. If your shop is committed to Oracle, can finance the licenses, and has the in-house talent -- use it. Nobody ever lost their job by buying IBM... -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR messed up when using queue
Hello list, I'm using an IVR where the caller chooses between 1. sales 2. support. When choosing 1 the caller is directed to the sales-queue when choosing 2 the caller is directed to the support-queue. Then the caller is directed to a free agent. I notice in the CDR-rapports that the destination is always '1' or '2', namely the DTMF-digit that the caller presses. How can I get here the name of the agent or its number as final destination of the call ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting to an External EPBX without an SIP provider
Hi, If I get a Dignum Card and fit it into my computer do I still need an SIP provider to connect through my EPBX to a Public Telephone System? Thanks --Siju -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR messed up when using queue
from queue.conf ; UpdateCDR behavior. ;This option is implemented to mimic chan_agents behavior of populating ;CDR dstchannel field of a call with an agent name, which you can set ;at the login time with AddQueueMember membername parameter. ; ; updatecdr = no I've never used it. YMMV Alec _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Wednesday, 27 January 2010 10:25 p.m. To: Asterisk Mailing Subject: [asterisk-users] CDR messed up when using queue Hello list, I'm using an IVR where the caller chooses between 1. sales 2. support. When choosing 1 the caller is directed to the sales-queue when choosing 2 the caller is directed to the support-queue. Then the caller is directed to a free agent. I notice in the CDR-rapports that the destination is always '1' or '2', namely the DTMF-digit that the caller presses. How can I get here the name of the agent or its number as final destination of the call ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unregistred users can pass calls, peer being static
Hi, we had an attack on a server and we don't understand how it was possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL, network 188.161.128.0/18 Hacked account had following setup: [111] type=friend username=111 context=from-111 host=11.22.33.44 dtmfmode=auto qualify=yes nat=yes canreinvite=no defaultip=11.22.33.44 port=35060 disallow=all allow=ulaw,alaw call-limit=2 Despite this, I saw in my logs that someone hacked this account and could place calls! in logs we have: [Jan 27 04:00:13] ERROR[29715] chan_sip.c: Peer '111' is trying to register, but not configured as host=dynamic [Jan 27 04:00:13] NOTICE[29715] chan_sip.c: Registration from 'sip:1...@ourasteriskip' failed for '188.161.152.245' - Peer is not supposed to register [Jan 27 04:00:18] VERBOSE[30669] logger.c: -- Executing [972599400...@from-111:1] NoOp(SIP/111-16eb, Incoming call from ) in new stack As you see 111 could place a call even having not registered, which he is not supposed to do. How is this possible? -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR messed up when using queue
I'm using Asterisk 1.4.27. In queues.conf I do not find this option. I have added it, reloaded Asterisk, but still the destination is '1' or '2'. Does it make a difference of my queue members are just SIP-accounts in stead of agents ? member = SIP/VCsupport,1,Jonas member = Agent/VCjoeri,2,Joeri Jonas. On Wed, 2010-01-27 at 23:13 +1300, Alec Davis wrote: from queue.conf ; UpdateCDR behavior. ;This option is implemented to mimic chan_agents behavior of populating ;CDR dstchannel field of a call with an agent name, which you can set ;at the login time with AddQueueMember membername parameter. ; ; updatecdr = no I've never used it. YMMV Alec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detected digit 'f' - SOLVED
On Tue, 2010-01-26 at 13:17 -0600, Kevin P. Fleming wrote: Jeff Brower wrote: How do you know for sure fax detection is turned off? It sounds to me like your changes to the dahdi config file are being ignored. Maybe put something in there that should cause an error or something clearly observable, then see whether that actually occurs. Or even easier, use 'dahdi show channel X' and see if faxdetect is indeed disabled. Hi, Thanks for the tip - I've learnt a new Asterisk command :) According to that it was already off (output at bottom). However, something (can't remember what) made me look at the wanpipe config and I discovered that in the one that worked, I had this (I'm only showing the differences here): [wanpipe1] TDMV_HW_DTMF = NO [w1g1] TDMV_ECHO_OFF = NO Whereas on the new one that wasn't working with our old fax machine, there was this: [wanpipe1] TDMV_HW_DTMF = YES TDMV_HW_FAX_DETECT = YES [w1g1] (no entry for TDMV_ECHO_OFF) I copied the wanpipe config files over from the older working server and it fixed it :) FWIW we're using these cards: http://www.voipon.co.uk/sangoma-a104de-p-328.html Cheers, Kingsley. sc-gw1*CLI core show channels verbose Channel Context ExtensionPrio State Application Data CallerIDDuration Accountcode BridgedTo IAX2/iaxmodem0-7025 ext-fax 1 Up AppDial (Outgoing Line) 08454632501 00:00:31 DAHDI/34-1 DAHDI/34-1 recvfax 08454632501 7 Up Dial IAX2/iaxmodem0/0845463250 1276459900 00:00:36 IAX2/iaxmodem0-7025 2 active channels 1 active call sc-gw1*CLI dahdi show channel 34-1 Channel: 34 File Descriptor: 52 Span: 2 Extension: 08454632501 Dialing: no Context: zapinbound Caller ID: 1276459900 Calling TON: 33 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: ISDN PRI Radio: 0 Owner: DAHDI/34-1 Real: DAHDI/34-1 Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: yes Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently ON PRI Flags: Call PRI Logical Span: Implicit Actual Confinfo: Num/0, Mode/0x Actual Confmute: Yes Hookstate (FXS only): Onhook sc-gw1*CLI core show channel IAX2/iaxmodem0-7025 -- General -- Name: IAX2/iaxmodem0-7025 Type: IAX2 UniqueID: 1264584800.13 Caller ID: 08454632501 Caller ID Name: (N/A) DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormats: 0x8 (alaw) WriteFormat: 0x8 (alaw) ReadFormat: 0x8 (alaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: -1 Frames in: 1136 Frames out: 1 Time to Hangup: 0 Elapsed Time: 0h0m23s Direct Bridge: DAHDI/34-1 Indirect Bridge: DAHDI/34-1 -- PBX -- Context: ext-fax Extension: Priority: 1 Call Group: 0 Pickup Group: 0 Application: AppDial Data: (Outgoing Line) Blocking in: ast_waitfor_nandfds Variables: BRIDGEPEER=DAHDI/34-1 DIALEDPEERNUMBER=iaxmodem0/08454632501 FAXFILE=/var/spool/asterisk/fax/20100127093315-1276459900-08454632501 item_count=1 fax2email_from_name=Fax fax2email_dni=08454632501 fax2email_cli=441276459900 channel1=1264584795.12 destination_type_id_1=7 fax2email_from_addr=fax2em...@skycomuk.com fax2email_paper_size=a4 fax2email_format=pdf fax2email_to_addr=kings...@noodles.skymarket.co.uk recordingemailto=kingsley.t...@gmail.com recordingemailfrom= recordcall=1 dniascli=0 custid=13361 app=service_nts_nextgen_v2 CDR Variables: level 1: dst=s level 1: dcontext=default level 1: channel=IAX2/iaxmodem0-7025 level 1: start=2010-01-27 09:33:20 level 1: answer=2010-01-27 09:33:20 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1264584800.13 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] astdb
Hi, all What is the use of astdb? Is it used to store realtime values like sip etc. Regards, Bhrugu Mehta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attended Transfer with REFER
Thanks a lot guys. Exactly what I needed. Best regards, Örn On Tue, Jan 26, 2010 at 8:48 PM, Olle E. Johansson o...@edvina.net wrote: 26 jan 2010 kl. 16.48 skrev Örn Arnarson: Hi guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a timeout in the Dial command is not suitable unless I am able to tell somehow that the call in question is being forwarded (which is of course not the case, as the Dial command is called befer the REFER is sent). Can anyone think of a way to get the call back to the transferrer after this timeout? THe transferred call is sent to a context set with the channel variable TRANSFER_CONTEXT before you call DIAL(). In there, run DUMPCHAN to see which variables you have and then dial with a timeout. After the timeout, dial back. /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregistred users can pass calls, peer being static
On Wed, 2010-01-27 at 11:47 +0100, Administrator TOOTAI wrote: Hi, we had an attack on a server and we don't understand how it was possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL, network 188.161.128.0/18 Hacked account had following setup: [111] type=friend username=111 context=from-111 host=11.22.33.44 dtmfmode=auto qualify=yes nat=yes canreinvite=no defaultip=11.22.33.44 port=35060 disallow=all allow=ulaw,alaw call-limit=2 Despite this, I saw in my logs that someone hacked this account and could place calls! in logs we have: [Jan 27 04:00:13] ERROR[29715] chan_sip.c: Peer '111' is trying to register, but not configured as host=dynamic [Jan 27 04:00:13] NOTICE[29715] chan_sip.c: Registration from 'sip:1...@ourasteriskip' failed for '188.161.152.245' - Peer is not supposed to register [Jan 27 04:00:18] VERBOSE[30669] logger.c: -- Executing [972599400...@from-111:1] NoOp(SIP/111-16eb, Incoming call from ) in new stack As you see 111 could place a call even having not registered, which he is not supposed to do. How is this possible? -- Daniel Check your sip.conf allowguest=no -- Best regards, Vince Mallow xmpp: w...@jabber.slan.ru web: http://gentoo-way.blogspot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregistred users can pass calls, peer being static
27 jan 2010 kl. 11.47 skrev Administrator TOOTAI: Hi, we had an attack on a server and we don't understand how it was possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL, network 188.161.128.0/18 Hacked account had following setup: [111] type=friend username=111 context=from-111 host=11.22.33.44 dtmfmode=auto qualify=yes nat=yes canreinvite=no defaultip=11.22.33.44 port=35060 disallow=all allow=ulaw,alaw call-limit=2 Despite this, I saw in my logs that someone hacked this account and could place calls! in logs we have: [Jan 27 04:00:13] ERROR[29715] chan_sip.c: Peer '111' is trying to register, but not configured as host=dynamic [Jan 27 04:00:13] NOTICE[29715] chan_sip.c: Registration from 'sip:1...@ourasteriskip' failed for '188.161.152.245' - Peer is not supposed to register [Jan 27 04:00:18] VERBOSE[30669] logger.c: -- Executing [972599400...@from-111:1] NoOp(SIP/111-16eb, Incoming call from ) in new stack As you see 111 could place a call even having not registered, which he is not supposed to do. How is this possible? Registration is a mechanism to tell the server where a phone can be reached when the phone wants to call it, thus registrations are only required for outbound calls. Inbound calls are not affected by registrations. type=friend creates two objects in your asterisk server, one peer and one user. Asterisk primarily match the user objects for incoming calls on the From: username. In this case, you have 111 as the username (regardless of the username field which is not the username btw). You have no secret defined, so anyone placing a call from a URI that has 111 as the username part will be able to use your server. Calling from sip:1...@asterisk.org as well as sip:1...@mydomain.com will work without authentication - from any IP address out there. Very poor security indeed. 1) Add a secret. 2) Add ACL rules (permit/deny) to restrict IP address access 3) Change to type=peer and we'll only match on IP for incoming calls. I still recommend using authentication. There has been a lot of information about how to secure your Asterisk on asterisk.org, this mailing list and in other forums. Make sure you read this and act upon it! Regards, /Olle -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom phone DND state
Hi, At my previous company we ran 1.4.x.x (underneath DiVitas.com software) and our Polycom IP 550 would use DND without a problem, but the IP 331 (on exactly the same server) didn't work with DND. So it may be a model-specific problem rather than your Asterisk config. Stuart From: Lee, John (Sydney) john@compuware.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wed, 27 January, 2010 8:02:14 Subject: Re: [asterisk-users] Polycom phone DND state I am using 1.4.21.2 and DND is definitely working. From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Saturday, 23 January 2010 2:50 AM To: ' Asterisk Users Mailing List - Non-Commercial Discussion ' Subject: [asterisk-users] Polycom phone DND state Hi, I know having Asterisk aware of Polycom Do No Disturb state wasn't working before (1.4), but is this working in any recent version? Is there any custom way of doing this? Regards, Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, NAT, and RTP?
Hello I think I finally understood the issue/solution, but I'd like to make sure I'm correct: - In Diana Cionoiu's famous article on Freshmeat (http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol), regardless of whether SIP end-points use a public IP or are behind a NAT, RTP packets flow directly between the two SIP end-points because the SIP server only acts... as an SIP server, meaning it only acts as a registrar (for SIP end-points to make themselves know with an IP + RTP ports), and then as a Central office (to ring the other SIP end-point, and close the connection when an SIP end-point decides to hangup) - OTOH, for IP PBX's like Asterisk to provide PBX services (eg. call transfer, call parking, etc.), it must remain in the loop, and hence, by default (canreinvite=no), all RTP packets always go through Asterisk, even if both SIP end-points live in the same network as the Asterisk server (and hence, since NAT is not involved, there's no need for any kung-fu with rewriting information in SDP packets and asking the NAT box to open the relevant ports for RTP) Is this correct? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR problems with Queue
Hi, I'm having problems with CDR's and Queues in Asterisk 1.6.1. Heres three examples: Normal call: User A calls in to asterisk, gets a PlayFile, and hangs up. This gives 1 CDR as expected. Call to a Queue and then a playfile afterwards: User A calls into asterisk, goes into a queue, asterisk calls user B. When user B hangs up a CDR for User A is generated. (no CDR for user B) User A continues to a PlayFile, and then hangs up. No CDR. Call directly to a Queue: User A calls into asterisk, goes into a queue, asterisk calls user B. When anyone hangs up, a CDR for User A is generated. No more CDR, both calls are hung up. 1. Why is CDR for user A written when user B hangs up, while A is still on the line? 2. Why isn't there any more CDRs for the call? Is there some special configuration I need to do, that I can't seem to find any documentation of? I tried searching for asterisk queue cdr on google, but I only get info about queue_log, which is not what I want. I want numbers and channels written as usual in the cdr log. Anyone have any ideas why this is happening, and what I can do? Håkon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astdb
Astdb is a built-in Berkley database that Asterisk uses via a specific command set. It is (IMO) simpler to use than MYSQL, POSTGRES or whatever other flavor of database you might use (odbc, etc). It does not (necessarily) store realtime values; it's more of a simple push/pull single key database. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bhrugu mehta Sent: Wednesday, January 27, 2010 5:07 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] astdb Hi, all What is the use of astdb? Is it used to store realtime values like sip etc. Regards, Bhrugu Mehta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR messed up when using queue
forkCDR might be helpful; also, you might want to check all of the CDR fields. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Wednesday, January 27, 2010 3:25 AM To: Asterisk Mailing Subject: [asterisk-users] CDR messed up when using queue Hello list, I'm using an IVR where the caller chooses between 1. sales 2. support. When choosing 1 the caller is directed to the sales-queue when choosing 2 the caller is directed to the support-queue. Then the caller is directed to a free agent. I notice in the CDR-rapports that the destination is always '1' or '2', namely the DTMF-digit that the caller presses. How can I get here the name of the agent or its number as final destination of the call ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR problems with Queue
Just a shot in the dark what is the endbeforehexten value in cdr.conf? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Håkon Nessjøen Sent: Wednesday, January 27, 2010 7:59 AM To: Asterisk Users Mailing List Subject: [asterisk-users] CDR problems with Queue Hi, I'm having problems with CDR's and Queues in Asterisk 1.6.1. Heres three examples: Normal call: User A calls in to asterisk, gets a PlayFile, and hangs up. This gives 1 CDR as expected. Call to a Queue and then a playfile afterwards: User A calls into asterisk, goes into a queue, asterisk calls user B. When user B hangs up a CDR for User A is generated. (no CDR for user B) User A continues to a PlayFile, and then hangs up. No CDR. Call directly to a Queue: User A calls into asterisk, goes into a queue, asterisk calls user B. When anyone hangs up, a CDR for User A is generated. No more CDR, both calls are hung up. 1. Why is CDR for user A written when user B hangs up, while A is still on the line? 2. Why isn't there any more CDRs for the call? Is there some special configuration I need to do, that I can't seem to find any documentation of? I tried searching for asterisk queue cdr on google, but I only get info about queue_log, which is not what I want. I want numbers and channels written as usual in the cdr log. Anyone have any ideas why this is happening, and what I can do? Håkon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregistred users can pass calls, peer being static
wins mallow a écrit : On Wed, 2010-01-27 at 11:47 +0100, Administrator TOOTAI wrote: [...] Check your sip.conf allowguest=no Guest are allowed and going to a different context. Logs are showing that calls are going out to the from-111 context, so its this account which was hacked. Thanks for your answer. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR problems with Queue
On Wed, Jan 27, 2010 at 3:37 PM, Danny Nicholas da...@debsinc.com wrote: Just a shot in the dark – what is the endbeforehexten value in cdr.conf? It was not defined. And the result was the same either it was set to yes or no. The cdr closing when user B hangs up, gets the full duration of call A at that point. And no more cdrs exists. Håkon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregistred users can pass calls, peer being static
Olle E. Johansson a écrit : 27 jan 2010 kl. 11.47 skrev Administrator TOOTAI: Hi, we had an attack on a server and we don't understand how it was possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL, network 188.161.128.0/18 Hacked account had following setup: [111] type=friend username=111 context=from-111 host=11.22.33.44 dtmfmode=auto qualify=yes nat=yes canreinvite=no defaultip=11.22.33.44 port=35060 disallow=all allow=ulaw,alaw call-limit=2 Despite this, I saw in my logs that someone hacked this account and could place calls! in logs we have: [Jan 27 04:00:13] ERROR[29715] chan_sip.c: Peer '111' is trying to register, but not configured as host=dynamic [Jan 27 04:00:13] NOTICE[29715] chan_sip.c: Registration from 'sip:1...@ourasteriskip' failed for '188.161.152.245' - Peer is not supposed to register [Jan 27 04:00:18] VERBOSE[30669] logger.c: -- Executing [972599400...@from-111:1] NoOp(SIP/111-16eb, Incoming call from ) in new stack As you see 111 could place a call even having not registered, which he is not supposed to do. How is this possible? [...] type=friend creates two objects in your asterisk server, one peer and one user. Asterisk primarily match the user objects for incoming calls on the From: username. In this case, you have 111 as the username (regardless of the username field which is not the username btw). You have no secret defined, so anyone placing a call from a URI that has 111 as the username part will be able to use your server. Calling from sip:1...@asterisk.org as well as sip:1...@mydomain.com will work without authentication - from any IP address out there. Very poor security indeed. 1) Add a secret. 2) Add ACL rules (permit/deny) to restrict IP address access 3) Change to type=peer and we'll only match on IP for incoming calls. I still recommend using authentication. So the fact that host is setted to an IP doesn't matter in case of type=friend. Didn't notice that, thanks for the explanation. [..] Make sure you read this and act upon it! Sure, already done. Thanks for your answer. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting to an External EPBX without an SIP provider
In this case, a SIP provider would not be required. Obviously, you will need ports on your EPBX to connect the Digium card to. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siju George Sent: Wednesday, January 27, 2010 5:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Connecting to an External EPBX without an SIP provider Hi, If I get a Dignum Card and fit it into my computer do I still need an SIP provider to connect through my EPBX to a Public Telephone System? Thanks --Siju -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom vs Polycom
On Mon, Jan 25, 2010 at 12:07:55PM -0600, Karl Fife wrote: From: cb c...@mythtech.net Sent: Sunday, January 24, 2010 12:42 I use the Snom 370 all day long at work. I have never had a problem adjusting the volume. I change it multiple times a day as I keep my handset on one volume and my headset on another, so I'm always going up and down and I've never accidentally pressed any other key. I will however agree with you on the Mute button, any time I want to mute a call, I have to stop and look at the buttons and figure out which one of the tiny ones is mute. You are right, especially with practice. I still think it's a little easier if the buttons are distinct from one another by appearance, size and/or placement.. I think Polycom made a good design decision by not making you 'reach over' any buttons to press these common buttons. I also like the fact that the mute button turns bright red when activated. Come to think of it, I wish the DND button turned red when activated :-). ... I have a giga-bit and wifi phone... Tzafrir Cohen LOL. My phone? It's a large white box with seven PCIx/e slots running RAID 1 sporting a 3-screen monitor tree :-) -Karl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
On Wednesday 27 January 2010 01:34:57 Zhang Shukun wrote: 2010/1/23 Steve Edwards asterisk@sedwards.com: On Fri, 22 Jan 2010, Zhang Shukun wrote: as you know, we can use MYSQL command to visit mysql database but if i use other database like Oracke,sybase,etc, Could i use MYSQL command ? ODBC will do what you want. Thanks, while i think because oracle has no offical ODBC for linux system. I've used this driver with Oracle before with good results. Just make sure that you use the full Oracle libraries, not the Instant Client. The Instant Client has a severe resource leak that makes it inappropriate for long-running processes. http://home.fnal.gov/~dbox/oracle/odbc/ -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GoToIfTime issue
On Wednesday 27 January 2010 01:48:47 Zhang Shukun wrote: how does the system recognize them. i mean queue_name is not an configure option in agent.conf The name between the square brackets in queues.conf is the queue_name. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need recommendation for ISDN-BRI cards for use with Asterisk
Hi, I have very limited experience with BRIs, and now I have a project which requires to hook up an asterisk server to a client's Simen Hicom PBX with 32 BRI ports. In this regard I am looking for the right ISDN-BRI cards which I can install in an Asterisk server. I need two types of cards, one which could support 2-wire U2b1 interface and two which could support 4-wire S0 interface. Kindly guide me in the right direction for these cards. Thanks, -- Zeeshan A Zakaria -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
This is WAY OT but I had no idea what fnal.gov was, so I checked it out: http://computing.fnal.gov/xms/Services/Getting_Services/Web_at_Fermilab/Professional_Home_Pages_at_Fermilab And I quote ...professional information about themself... About themself? Really? Really? That is all. Cheers Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Wednesday, January 27, 2010 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MYSQL problem On Wednesday 27 January 2010 01:34:57 Zhang Shukun wrote: 2010/1/23 Steve Edwards asterisk@sedwards.com: On Fri, 22 Jan 2010, Zhang Shukun wrote: as you know, we can use MYSQL command to visit mysql database but if i use other database like Oracke,sybase,etc, Could i use MYSQL command ? ODBC will do what you want. Thanks, while i think because oracle has no offical ODBC for linux system. I've used this driver with Oracle before with good results. Just make sure that you use the full Oracle libraries, not the Instant Client. The Instant Client has a severe resource leak that makes it inappropriate for long-running processes. http://home.fnal.gov/~dbox/oracle/odbc/ -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR messed up when using queue
Hello Danny, what do you mean by 'all the CDR fields' ? The Destination-field shows '1' or '2'. The dstchannel shows the correct SIP-channel. But this is not the same as the 'real' destination namely the SIP-account of my SIP-phone. Jonas. On Wed, 2010-01-27 at 08:22 -0600, Danny Nicholas wrote: forkCDR might be helpful; also, you might want to check all of the CDR fields. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mitel integration
Hi, A potential client (hotel) has a Property Management System that talks the Mitel protocol to their current Mitel PBX in order to receive CDRs (which end up being rated by the PMS system and charged back to guests). Does anyone know of any (free or otherwise) docs on this protocol, or better still have experience interfacing asterisk in a hotel situation like this? The PMS developers claim that the Mitel spec is proprietary, and that they cannot give it to me, and are basically unwilling to try and develop a method with us to integrate directly. Funny enough they also claim that just about every traditional PBX emulates this protocol for integration with PMS systems, so they say that if I can manage to do the same I will instantly integrate with MANY PMS systems. Sounds good to me, but without the spec I'm stuck in a catch 22! Thanks, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mitel integration
On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote: Sounds good to me, but without the spec I'm stuck in a catch 22! tcpdump? (assuming IP). Bet its fairly simple plain text or something. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mitel integration
the mitel 3300 sends SMDR on TCP 1752. It spews software and hardware logs in the same manner, different ports. On 1/27/2010 11:00 AM, Steve Howes said: On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote: Sounds good to me, but without the spec I'm stuck in a catch 22! tcpdump? (assuming IP). Bet its fairly simple plain text or something. Steve attachment: Mark_Wiater.vcf-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Database Configuration
Hello
I need to add sip extensions from my UI so without going through sip.conf so
i created table
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`username` varchar(40) default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
`ipaddr` varchar(20) NOT NULL default '',
`port` int(6) NOT NULL default '0',
`regseconds` int(11) NOT NULL default '0',
PRIMARY KEY (`name`)
) TYPE=MyISAM
then i put sip.conf
[general]
hostname=localhost
dbname=asterisk
table= sipfriends
password=ahmed
user=root
then i insert in sql this statment insert into sipfriends values
('555','555','1234','555','192.168.50.149',5060,2);
i tried from Xlite to register with 555 but i couldn't
any help please
--
Ahmed Magdy Mahmoud
--
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Re: [asterisk-users] Mitel integration
On Wed, 27 Jan 2010, Mark Wiater wrote: the mitel 3300 sends SMDR on TCP 1752. It spews software and hardware logs in the same manner, different ports. This particular model (need to get the model number) has a serial connection. I'm all for putting a serial sniffer between them (if they let me!), but was really hoping someone had already done this and could give me a headstart. I'll investigate the ethernet options, though, as that would make more sense anyway! If the PMS will talk over ethernet I'll try to pretend to be a 3300. Cheers, j On 1/27/2010 11:00 AM, Steve Howes said: On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote: Sounds good to me, but without the spec I'm stuck in a catch 22! tcpdump? (assuming IP). Bet its fairly simple plain text or something. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregistred users can pass calls, peer being static
Administrator TOOTAI wrote: Olle E. Johansson a écrit : 27 jan 2010 kl. 11.47 skrev Administrator TOOTAI: Hi, we had an attack on a server and we don't understand how it was possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL, network 188.161.128.0/18 Hacked account had following setup: [111] type=friend username=111 context=from-111 host=11.22.33.44 dtmfmode=auto qualify=yes nat=yes canreinvite=no defaultip=11.22.33.44 port=35060 disallow=all allow=ulaw,alaw call-limit=2 Despite this, I saw in my logs that someone hacked this account and could place calls! in logs we have: [Jan 27 04:00:13] ERROR[29715] chan_sip.c: Peer '111' is trying to register, but not configured as host=dynamic [Jan 27 04:00:13] NOTICE[29715] chan_sip.c: Registration from 'sip:1...@ourasteriskip' failed for '188.161.152.245' - Peer is not supposed to register [Jan 27 04:00:18] VERBOSE[30669] logger.c: -- Executing [972599400...@from-111:1] NoOp(SIP/111-16eb, Incoming call from ) in new stack As you see 111 could place a call even having not registered, which he is not supposed to do. How is this possible? [...] type=friend creates two objects in your asterisk server, one peer and one user. Asterisk primarily match the user objects for incoming calls on the From: username. In this case, you have 111 as the username (regardless of the username field which is not the username btw). You have no secret defined, so anyone placing a call from a URI that has 111 as the username part will be able to use your server. Calling from sip:1...@asterisk.org as well as sip:1...@mydomain.com will work without authentication - from any IP address out there. Very poor security indeed. 1) Add a secret. 2) Add ACL rules (permit/deny) to restrict IP address access 3) Change to type=peer and we'll only match on IP for incoming calls. I still recommend using authentication. So the fact that host is setted to an IP doesn't matter in case of type=friend. Didn't notice that, thanks for the explanation. [..] Make sure you read this and act upon it! This conversation brings to mind two possible ways we could improve Asterisk to help users from falling into this trap: 1) When a sip.conf entry is defined as 'type=friend' *and* has a specific host IP address (not dynamic), we could just ignore the 'user' part and create only the 'peer' part. This would result in incoming calls being matched by IP address instead of username, which is likely what the administrator wants anyway. 2) Alternatively, if people really do want both the 'user' and 'peer' objects to exist, then we could automatically put an ACL on the 'user' object that restricts access to it to only the defined IP address. This also could apply to dynamic hosts, but only those that are defined without a secret (no authentication required), which seems like a terrible configuration and we don't really need to do anything to make it work 'better' :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Queue not work in 1.6.2.1
On Wed, 2010-01-27 at 10:27 +0800, Zhang Shukun wrote: hi,all i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except realtime queue. it seems queue_table works fine, but queue_member_queue not work, the two tables works fine when in 1.4.28. is that something changed related to realtime queue configuration? more detail about two table definition and data stored in , please see: http://pastebin.com/m33f9539e the extconfig.conf file, please see: http://pastebin.com/m2008ced1 and the res_mysql.conf file: http://pastebin.com/m27d3fdc5 Could you tell me what's wrong with me ? Thanks! How do your agents log into the system? If you were using agentcallbacklogin that was deprecated and does not exist in version 1.6 of Asterisk. The queue_member_table was used by agentcallbacklogin or the agentlogin commands. With Asterisk 1.6 you are supposed to be using dynamic agents so there is no purpose for that table. That is what may be wrong with Asterisk. What is wrong with you is a very different question ;) -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Connected to definity errors
Did you get this resolved? And how if you did. We've been have the same random PRI lockup issue for years now. I've opened a mantis bug https://issues.asterisk.org/view.php?id=16713 and hopefully we can get this issue resolved. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Thursday, 20 August 2009 11:21 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PRI Connected to definity errors We have setup asterisk to handle our calls before between telco and an Avaya definity. The PRI keeps locking up every so often. In addition I keep getting this error when trying to call the avaya: -- Channel 0/2, span 1 got hangup request, cause 102 -- Hungup 'Zap/2-1' When that error happens I get a fast busy (congestion) tone. Any one can point me in the right direction? TIA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregistred users can pass calls, peer being static
On Wed, Jan 27, 2010 at 6:10 PM, Kevin P. Fleming kpflem...@digium.comwrote: 1) When a sip.conf entry is defined as 'type=friend' *and* has a specific host IP address (not dynamic), we could just ignore the 'user' part and create only the 'peer' part. This would result in incoming calls being matched by IP address instead of username, which is likely what the administrator wants anyway. I think it would make more sense to give a warning about illegal use of the host parameter when type=friend. This way the user gets information of why this is wrong, instead of continuing to misuse the parameter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
2010/1/27 Steve Edwards asterisk@sedwards.com: So don't use ODBC, use Pro*C... On Wed, 27 Jan 2010, Zhang Shukun wrote: you said Personally, I'd vote for an AGI using whatever C API your DB provides what do you think about phpagi and cagi, if i choose the agi method. while phpagi seems used more popular than cagi. I write my AGIs in C because that's my sharpest tool and because I'm old-school enough to care about shaving milliseconds. You can execute xxx AGIs written in C in the time it will take to load the PHP interpreter and parse your script. If you expect to process a lot of calls, all those milliseconds add up -- at least in my mind. I wrote my own AGI library because I started before cagi was available so I have no experience with it. On Wed, 27 Jan 2010, Zhang Shukun wrote: i cant quantify my requirement now. but the business has not be start. and the user will increase as time going. If you don't know where you are going how will you know what road to take? If you and your peers don't know how big your database will be or how many queries per second how can you say MySQL is not up to the job? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
snip many people around think mysql is not a good option for database, they think mysql is only suit for small business. but i want to have a try. i need to convince them to use this. /snip This statement is absolute BS. Give me some factual, backed statements by trained database professionals who don't work for Microsoft or Oracle (OK, Sun) and we can talk. rant I guarantee that mysql can be made at least as fast as Oracle with a relational database that's designed, indexed and implemented properly. The problem with the backend is NEARLY ALWAYS a problem with the DBA. I hate to hear crap DBAs blame their problems on the backend. MySQL is top-notch and production ready if you are logical about your DB design. /rant -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR problems with Queue
2010/1/27 Håkon Nessjøen haa...@avelia.no On Wed, Jan 27, 2010 at 3:37 PM, Danny Nicholas da...@debsinc.com wrote: Just a shot in the dark – what is the endbeforehexten value in cdr.conf? It was not defined. And the result was the same either it was set to yes or no. The cdr closing when user B hangs up, gets the full duration of call A at that point. And no more cdrs exists. Håkon Anyone? :/ Do all of you get two CDR's in calls to queues? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR problems with Queue
This stands to be corrected, but as I understand it, the queue command on its own would not generate a second CDR any more than a transfer to an extension. The way I understand the queue/agent/call relationship is this: 1. agent(s) login to queue this may or may not create a CDR entry 2. caller calls asterisk this creates a CDR for that incoming call 3. caller is sent to queue by context or dialplan this creates a CDR only if forkCDR is used since the queue command isnt a new call It seems to me that would be desirable for queue to act as a second pbx where queue activity (login/logout/transfer etc) is logged in the CDR as new calls, Since my queue experience and $5 will get you a decent starbucks in some places, take this with a grain of salt, please. -- Danny Nicholas -- _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Håkon Nessjøen Sent: Wednesday, January 27, 2010 1:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR problems with Queue 2010/1/27 Håkon Nessjøen haa...@avelia.no On Wed, Jan 27, 2010 at 3:37 PM, Danny Nicholas da...@debsinc.com wrote: Just a shot in the dark what is the endbeforehexten value in cdr.conf? It was not defined. And the result was the same either it was set to yes or no. The cdr closing when user B hangs up, gets the full duration of call A at that point. And no more cdrs exists. Håkon Anyone? :/ Do all of you get two CDR's in calls to queues? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mitel integration
Having looked at the outputs into PMS they are very simple stop start records. Line by line text that can easily be recreated. They have about 4-5 fields, origin number, destination, time of call, duration, or similar things Usually they go out via a serial port or TCP port expecting a terminal to receive them so plugging into them will quickly show you what you need. Its not that you need to match the Mitel, you need to match the PMS. Best to talk to them but I have looked at it for a couple of customers who are still deciding and helped them fix their PBXs when they broke and its pretty straight forward. You just need to be able to output to either a serial or TCP port . Cheers Duncan On 28/01/2010, at 6:01 AM, Jeff LaCoursiere wrote: On Wed, 27 Jan 2010, Mark Wiater wrote: the mitel 3300 sends SMDR on TCP 1752. It spews software and hardware logs in the same manner, different ports. This particular model (need to get the model number) has a serial connection. I'm all for putting a serial sniffer between them (if they let me!), but was really hoping someone had already done this and could give me a headstart. I'll investigate the ethernet options, though, as that would make more sense anyway! If the PMS will talk over ethernet I'll try to pretend to be a 3300. Cheers, j On 1/27/2010 11:00 AM, Steve Howes said: On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote: Sounds good to me, but without the spec I'm stuck in a catch 22! tcpdump? (assuming IP). Bet its fairly simple plain text or something. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Data transfer
Hi, im a student and we are devloping a training sytem for radio operators (for ships, police, ...) at our university. So far we are using a simple own protocol for speech and data transmission, works well at a Lan. Now we are looking for a way to connect the devices over the internet. I did some very quick testing with Asterisk and PJSIP [1] and it looks very promising. Apart from the voice transmission we need to sent some Data too (like used frequency, GPS position, very small data, about 5 kByte a minute). So my first thougt was to use SIP/Messages but some time of searching shows that asterisk doesn't handle this. We could of course use an extra tcp connection but this seems not very elegant to me ;-) because SIP should handle that... The Asterisk Console shows that asterisk drops the message: WARNING[15294]: chan_sip.c:9769 receive_message: Received message to sip:test.us...@192.168.1.104 from sip:192.168.1.101;tag=yNOCnaUdAjHob7Gmpl-5tjCuNmQDeGJp, dropped it... Content-Type:text/plain Message: gnaaa Is there a way to get this message out of the server - to the AMI Interface for example? This would enough for us, because only our server needs to read the messages and maybe sent an answer. Or someone has an even better idee how to achive this? Thank you, Thorsten Stoffregen [1] www.pjsip.org/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR problems with Queue
On Wed, Jan 27, 2010 at 8:22 PM, Danny Nicholas da...@debsinc.com wrote: This stands to be corrected, but as I understand it, the queue command on it’s own would not generate a second CDR any more than a transfer to an extension. The way I understand the queue/agent/call relationship is this: 1. agent(s) login to queue – this may or may not create a CDR entry 2. caller calls asterisk – this creates a CDR for that incoming call 3. caller is sent to queue by context or dialplan – this creates a CDR only if forkCDR is used since the queue command isn’t a “new call” It seems to me that would be “desirable” for queue to act as a “second pbx” where queue activity (login/logout/transfer etc) is logged in the CDR as new calls, Since my queue experience and $5 will get you a decent starbucks in some places, take this with a grain of salt, please. Hi, I'm not sure if you understand me correctly. I'm not interested in cdrs for agent logins. For me, that is not call detail. But when you call into the pbx, thats one leg, and one cdr. And when a queue() calls out to an agent, that is also a call. And for me, it's logical that this would be two cdr lines. Because if this is going in and out of a PRI/SS7 line, you would get money and pay money for these calls. And therefore it's very important that queue() would start the cdr duration a the exact moment the agent answers. etc. But the weirdest thing, is that the CDR for leg A is written when queue() is done, even if leg A is still active. I can't understand how people would accept this behaviour as logical? There must be a way to make queue() create valid cdrs? The one I get isn't correct for either legs. Regards, Håkon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Connected to definity errors
We didn't fix it yet. For the moment the Definity is not connected directly to Asterisk, we route all communications between Asterisk and the Definity over the PSTN. The plan is to play around with all protocol settings to figure out which one is the most stable, from what I understand - however I haven't yet tested it - att custom should work best. But we didn't yet get around to it. On Wed, Jan 27, 2010 at 12:47 PM, Alec Davis siva...@paradise.net.nz wrote: Did you get this resolved? And how if you did. We've been have the same random PRI lockup issue for years now. I've opened a mantis bug https://issues.asterisk.org/view.php?id=16713 and hopefully we can get this issue resolved. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Thursday, 20 August 2009 11:21 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PRI Connected to definity errors We have setup asterisk to handle our calls before between telco and an Avaya definity. The PRI keeps locking up every so often. In addition I keep getting this error when trying to call the avaya: -- Channel 0/2, span 1 got hangup request, cause 102 -- Hungup 'Zap/2-1' When that error happens I get a fast busy (congestion) tone. Any one can point me in the right direction? TIA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Database Configuration
Can you link the howto or other documentation you are following to set this up?
What version of asterisk?
Did you edit extconfig.conf?
Heres a howto for 1.4.x
http://hostseries.com/asterisk-realtime-installation-guide/
On Wed, Jan 27, 2010 at 8:39 AM, ahmed magdy amagdy.ibra...@gmail.com wrote:
Hello
I need to add sip extensions from my UI so without going through sip.conf so
i created table
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`username` varchar(40) default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
`ipaddr` varchar(20) NOT NULL default '',
`port` int(6) NOT NULL default '0',
`regseconds` int(11) NOT NULL default '0',
PRIMARY KEY (`name`)
) TYPE=MyISAM
then i put sip.conf
[general]
hostname=localhost
dbname=asterisk
table= sipfriends
password=ahmed
user=root
then i insert in sql this statment insert into sipfriends values
('555','555','1234','555','192.168.50.149',5060,2);
i tried from Xlite to register with 555 but i couldn't
any help please
--
Ahmed Magdy Mahmoud
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Re: [asterisk-users] Asterisk, NAT, and RTP?
You'd need RTP ports open for asterisk then. Transfers and parking can be done at the SIP level, asterisk doesn't have to be in the RTP path, as it can reinvite itself into the callpath as necessary. On Wed, Jan 27, 2010 at 5:23 AM, Vincent codecompl...@free.fr wrote: Hello I think I finally understood the issue/solution, but I'd like to make sure I'm correct: - In Diana Cionoiu's famous article on Freshmeat (http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol), regardless of whether SIP end-points use a public IP or are behind a NAT, RTP packets flow directly between the two SIP end-points because the SIP server only acts... as an SIP server, meaning it only acts as a registrar (for SIP end-points to make themselves know with an IP + RTP ports), and then as a Central office (to ring the other SIP end-point, and close the connection when an SIP end-point decides to hangup) - OTOH, for IP PBX's like Asterisk to provide PBX services (eg. call transfer, call parking, etc.), it must remain in the loop, and hence, by default (canreinvite=no), all RTP packets always go through Asterisk, even if both SIP end-points live in the same network as the Asterisk server (and hence, since NAT is not involved, there's no need for any kung-fu with rewriting information in SDP packets and asking the NAT box to open the relevant ports for RTP) Is this correct? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR problems with Queue
I wonder how this would work for you? - exten = 1000,1,ForkCDR - exten = 1000,2,Queue(blah) - exten = 1000,3,Hangup This should do 2 CDRs for each queued call. CDR 1 would be the DAHDI to Queue time, CDR 2 queue to hangup. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Håkon Nessjøen Sent: Wednesday, January 27, 2010 3:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR problems with Queue On Wed, Jan 27, 2010 at 8:22 PM, Danny Nicholas da...@debsinc.com wrote: This stands to be corrected, but as I understand it, the queue command on its own would not generate a second CDR any more than a transfer to an extension. The way I understand the queue/agent/call relationship is this: 1. agent(s) login to queue this may or may not create a CDR entry 2. caller calls asterisk this creates a CDR for that incoming call 3. caller is sent to queue by context or dialplan this creates a CDR only if forkCDR is used since the queue command isnt a new call It seems to me that would be desirable for queue to act as a second pbx where queue activity (login/logout/transfer etc) is logged in the CDR as new calls, Since my queue experience and $5 will get you a decent starbucks in some places, take this with a grain of salt, please. Hi, I'm not sure if you understand me correctly. I'm not interested in cdrs for agent logins. For me, that is not call detail. But when you call into the pbx, thats one leg, and one cdr. And when a queue() calls out to an agent, that is also a call. And for me, it's logical that this would be two cdr lines. Because if this is going in and out of a PRI/SS7 line, you would get money and pay money for these calls. And therefore it's very important that queue() would start the cdr duration a the exact moment the agent answers. etc. But the weirdest thing, is that the CDR for leg A is written when queue() is done, even if leg A is still active. I can't understand how people would accept this behaviour as logical? There must be a way to make queue() create valid cdrs? The one I get isn't correct for either legs. Regards, Håkon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR problems with Queue
On Thu, Jan 28, 2010 at 12:03 AM, Danny Nicholas da...@debsinc.com wrote: I wonder how this would work for you? - exten = 1000,1,ForkCDR - exten = 1000,2,Queue(blah) - exten = 1000,3,Hangup This should do 2 CDR’s for each queued call. CDR 1 would be the DAHDI to Queue time, CDR 2 queue to hangup. CDR 1 will then still stop too early. CDR2 does not have callerid/channel for agent, and will probably have a duration from the ForkCDR() time, instead of from when the agent answers. :/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Connected to definity errors
Definity what? G3? I did that once, a real pain but doable. I don't remember the settings but if I had a terminal in front of me, I am sure I could get it work. Thanks, Steve T On Wed, Jan 27, 2010 at 5:42 PM, C F shma...@gmail.com wrote: We didn't fix it yet. For the moment the Definity is not connected directly to Asterisk, we route all communications between Asterisk and the Definity over the PSTN. The plan is to play around with all protocol settings to figure out which one is the most stable, from what I understand - however I haven't yet tested it - att custom should work best. But we didn't yet get around to it. On Wed, Jan 27, 2010 at 12:47 PM, Alec Davis siva...@paradise.net.nz wrote: Did you get this resolved? And how if you did. We've been have the same random PRI lockup issue for years now. I've opened a mantis bug https://issues.asterisk.org/view.php?id=16713 and hopefully we can get this issue resolved. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Thursday, 20 August 2009 11:21 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PRI Connected to definity errors We have setup asterisk to handle our calls before between telco and an Avaya definity. The PRI keeps locking up every so often. In addition I keep getting this error when trying to call the avaya: -- Channel 0/2, span 1 got hangup request, cause 102 -- Hungup 'Zap/2-1' When that error happens I get a fast busy (congestion) tone. Any one can point me in the right direction? TIA ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unregistred users can pass calls, peer being static
Hi Kevin Kevin P. Fleming a écrit : [...] This conversation brings to mind two possible ways we could improve Asterisk to help users from falling into this trap: 1) When a sip.conf entry is defined as 'type=friend' *and* has a specific host IP address (not dynamic), we could just ignore the 'user' part and create only the 'peer' part. This would result in incoming calls being matched by IP address instead of username, which is likely what the administrator wants anyway. 2) Alternatively, if people really do want both the 'user' and 'peer' objects to exist, then we could automatically put an ACL on the 'user' object that restricts access to it to only the defined IP address. This also could apply to dynamic hosts, but only those that are defined without a secret (no authentication required), which seems like a terrible configuration and we don't really need to do anything to make it work 'better' :-) #1 sounds great for me. Don't know for others but for us SIP EP are mainly setted as user host=dynamic+secret or host=IP address meaning permit only this IP. Other solution would be -in case of host=IP address- to set permit=IP address/32 deny=0.0.0.0/0.0.0.0 if those parameters are *not* present All of those solution are compatible with the fact that information should be given if the case appear. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL problem
On Wednesday 27 January 2010 12:55:18 David Gibbons wrote: snip many people around think mysql is not a good option for database, they think mysql is only suit for small business. but i want to have a try. i need to convince them to use this. /snip This statement is absolute BS. Give me some factual, backed statements by trained database professionals who don't work for Microsoft or Oracle (OK, Sun) and we can talk. rant I guarantee that mysql can be made at least as fast as Oracle with a relational database that's designed, indexed and implemented properly. The problem with the backend is NEARLY ALWAYS a problem with the DBA. I hate to hear crap DBAs blame their problems on the backend. MySQL is top-notch and production ready if you are logical about your DB design. /rant For OLTP, I'd agree with you, and in fact, I'd go one step further. Nothing can touch MySQL for speed with OLTP. However, if you're going to be doing massive joins for reporting, you're better off using something else (or running individual MySQL slaves, whose purpose is to run those complex queries and doing nothing else). In a past life, our MySQL database ran circles around Oracle, Informix, and DB2... until someone ran a massive join on the same server, which caused MySQL to crawl. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Data transfer
On Wednesday 27 January 2010 15:18:41 thorsten.stoffre...@gmx.de wrote: Hi, im a student and we are devloping a training sytem for radio operators (for ships, police, ...) at our university. So far we are using a simple own protocol for speech and data transmission, works well at a Lan. Now we are looking for a way to connect the devices over the internet. I did some very quick testing with Asterisk and PJSIP [1] and it looks very promising. Apart from the voice transmission we need to sent some Data too (like used frequency, GPS position, very small data, about 5 kByte a minute). So my first thougt was to use SIP/Messages but some time of searching shows that asterisk doesn't handle this. We could of course use an extra tcp connection but this seems not very elegant to me ;-) because SIP should handle that... The Asterisk Console shows that asterisk drops the message: WARNING[15294]: chan_sip.c:9769 receive_message: Received message to sip:test.us...@192.168.1.104 from sip:192.168.1.101;tag=yNOCnaUdAjHob7Gmpl-5tjCuNmQDeGJp, dropped it... Content-Type:text/plain Message: gnaaa Is there a way to get this message out of the server - to the AMI Interface for example? This would enough for us, because only our server needs to read the messages and maybe sent an answer. Or someone has an even better idee how to achive this? That's not exactly true. Asterisk merely requires that a call be up in order to pass text messages. It does not, however, allow text messages to be passed stateless. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Connected to definity errors
Alec Davis wrote: Did you get this resolved? And how if you did. We've been have the same random PRI lockup issue for years now. Really? We have a 1.4.x box hooked up directly to our Definity G3R via a PRI and a TN464F. I have yet to experience any PRI issues (That I'm aware of) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] yum install asterisk16 for Fedora Core 8
Hi Guys, I have tested and isntalled Asterisk 1.6.2 with FreePBX from Digium repos based on this url: http://www.asterisk.org/downloads/yum BUT that doesn't seem to work with Fedora instance which I am running on Amazon Ec2. Apparently Asterisk 1.4 is natively included in Fedora repository but not Asterisk 1.6. And when I added the Digium repository, it give me a 404 not found. I check and there is an RHEL folder but is empty of any RPMs vs CENTOS directory which has the rpms. Does this mean I have to install Asterisk from source on Fedora (which is definitly a pain)? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Queue not work in 1.6.2.1
2010/1/28 Carlos Chavez cur...@telecomabmex.com: On Wed, 2010-01-27 at 10:27 +0800, Zhang Shukun wrote: hi,all i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except realtime queue. it seems queue_table works fine, but queue_member_queue not work, the two tables works fine when in 1.4.28. is that something changed related to realtime queue configuration? more detail about two table definition and data stored in , please see: http://pastebin.com/m33f9539e the extconfig.conf file, please see: http://pastebin.com/m2008ced1 and the res_mysql.conf file: http://pastebin.com/m27d3fdc5 Could you tell me what's wrong with me ? Thanks! How do your agents log into the system? Thanks! i don't want to use agents member to login to system. i just want to set static SIP peers in the queue and they all can work according to the strategy when have call to the queue.just like follows: mysql select * from queue_table; +--+---+-+ | name | beginworktime | endworktime | +--+---+-+ | 950401234561 | 09:30:00 | 17:30:00| +--+---+-+ 3 rows in set (0.00 sec) mysql select * from queue_member_table; +--++--+---+-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++--+---+-++ | 18 | Zhang Shukun | 950401234561 | SIP/1001 | 0 | 1 | | 19 | Li Aiwei | 950401234561 | SIP/1002 | 0 | 1 | | 20 | Zhang Jianming | 950401234561 | SIP/1003 | 0 | 1 | +--++--+---+-++ 3 rows in set (0.00 sec) in above two table. queue:950401234561 have three queue members: SIP/1001 , SIP/1002 , SIP/1003 when Queue(950401234561) app is invoked, all three queue members will ring at the same time by default strategy(ringall). my problem now use asterisk 1.6.2.1 is : when Queue(950401234561) app is running, i can here music on hold, but none of my sip phones(SIP/1001 , SIP/1002 , SIP/1003) will ring, is that in asterisk 1.6.2.1, it's not support static realtime queue member any more? If you were using agentcallbacklogin that was deprecated and does not exist in version 1.6 of Asterisk. The queue_member_table was used by agentcallbacklogin or the agentlogin commands. With Asterisk 1.6 you are supposed to be using dynamic agents so there is no purpose for that table. That is what may be wrong with Asterisk. What is wrong with you is a very different question ;) -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom vs Polycom
If the computer is the same as the phone, one can't whine about breaking one while talking on the other :) On Wed, Jan 27, 2010 at 7:09 AM, Karl Fife karlf...@gmail.com wrote: On Mon, Jan 25, 2010 at 12:07:55PM -0600, Karl Fife wrote: From: cb c...@mythtech.net Sent: Sunday, January 24, 2010 12:42 I use the Snom 370 all day long at work. I have never had a problem adjusting the volume. I change it multiple times a day as I keep my handset on one volume and my headset on another, so I'm always going up and down and I've never accidentally pressed any other key. I will however agree with you on the Mute button, any time I want to mute a call, I have to stop and look at the buttons and figure out which one of the tiny ones is mute. You are right, especially with practice. I still think it's a little easier if the buttons are distinct from one another by appearance, size and/or placement.. I think Polycom made a good design decision by not making you 'reach over' any buttons to press these common buttons. I also like the fact that the mute button turns bright red when activated. Come to think of it, I wish the DND button turned red when activated :-). ... I have a giga-bit and wifi phone... Tzafrir Cohen LOL. My phone? It's a large white box with seven PCIx/e slots running RAID 1 sporting a 3-screen monitor tree :-) -Karl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linux-based hard phones?
Just wondering if there are any Linux-based hard phones out there -- if so, it'd be neat to see if I couldn't take advantage of the underlying OS. Thanks, -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting to an External EPBX without an SIP provider
Thanks for the reply jamie :-) Does ordinary EPBXs in US have those ports or do you need special EPBXs? --Siju On Wed, Jan 27, 2010 at 8:32 PM, Jamie A. Stapleton jstaple...@computer-business.com wrote: In this case, a SIP provider would not be required. Obviously, you will need ports on your EPBX to connect the Digium card to. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siju George Sent: Wednesday, January 27, 2010 5:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Connecting to an External EPBX without an SIP provider Hi, If I get a Dignum Card and fit it into my computer do I still need an SIP provider to connect through my EPBX to a Public Telephone System? Thanks --Siju -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astdb
hi, all thanks for reply, but actually i have configured sip to realtime and i got this message SIP Seeding peer from *astdb*: 'sip_ext' at sip_...@asterisk_ip:5060 for 60 so i have to know that my sip ext is stored in astdb or not. any other suggetion ? Regards, On Wed, Jan 27, 2010 at 4:37 PM, bhrugu mehta mehtabhr...@gmail.com wrote: Hi, all What is the use of astdb? Is it used to store realtime values like sip etc. Regards, Bhrugu Mehta -- Bhrugu Mehta Sr. S/W Engineer (DD) VOIP,Telephony Team India -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Database Configration
Hello
I need to add sip extensions from my UI so without going through sip.conf so
i created table
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`username` varchar(40) default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
`ipaddr` varchar(20) NOT NULL default '',
`port` int(6) NOT NULL default '0',
`regseconds` int(11) NOT NULL default '0',
PRIMARY KEY (`name`)
) TYPE=MyISAM
then i put sip.conf
[general]
hostname=localhost
dbname=asterisk
table= sipfriends
password=ahmed
user=root
then i insert in sql this statment insert into sipfriends values
('555','555','1234','555','192.168.50.149',5060,2);
i tried from Xlite to register with 555 but i couldn't
any help please
--
Ahmed Magdy Mahmoud
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