Re: [asterisk-users] i have a probleme and my asterisk and ovh
sip show peer ovh * Name : ovh Secret : Set MD5Secret: Not set Context : entrant-ovh Subscr.Cont. : Not set Language : fr AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : Yes DTMFmode : auto Timer T1 : 500 Timer B : 32000 ToHost : sip.ovh.net Addr-IP : 91.121.129.17 Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Transport: UDP Def. Username: 0033972112355 SIP Options : (none) Codecs : 0x100 (g729) Codec Order : (g729:20) Auto-Framing : No 100 on REG : No Status : UNREACHABLE Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs --- Retransmitting #1 (no NAT) to 91.121.129.17:5060: OPTIONS sip:sip.ovh.net SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport Max-Forwards: 70 From: asterisk sip:aster...@172.20.1.1;tag=as1545fb99 To: sip:sip.ovh.net Contact: sip:aster...@172.20.1.1 Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.8 Date: Wed, 08 Apr 2009 05:57:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- Retransmitting #6 (no NAT) to 91.121.129.17:5060: REGISTER sip:91.121.129.17 SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK312a379b;rport Max-Forwards: 70 From: sip:0033972112...@91.121.129.17;tag=as16505dec To: sip:0033972112...@91.121.129.17 Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51 CSeq: 1465 REGISTER User-Agent: Asterisk PBX 1.6.0.8 Authorization: Digest username=0033972112355, realm=sip.ovh.net, algorithm=MD5, uri=sip:91.121.129.17, nonce=0019c92d503f745637b43af4264a11db, response=04e848af655d00e03d032d9a1c2fae09, opaque=001934772ef6ed5 Expires: 120 Contact: sip:0033972112...@172.20.1.1 Event: registration Content-Length: 0 --- Retransmitting #2 (no NAT) to 91.121.129.17:5060: OPTIONS sip:sip.ovh.net SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport Max-Forwards: 70 From: asterisk sip:aster...@172.20.1.1;tag=as1545fb99 To: sip:sip.ovh.net Contact: sip:aster...@172.20.1.1 Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.8 Date: Wed, 08 Apr 2009 05:57:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 91.121.129.17:5060: OPTIONS sip:sip.ovh.net SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport Max-Forwards: 70 From: asterisk sip:aster...@172.20.1.1;tag=as1545fb99 To: sip:sip.ovh.net Contact: sip:aster...@172.20.1.1 Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.8 Date: Wed, 08 Apr 2009 05:57:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (no NAT) to 91.121.129.17:5060: OPTIONS sip:sip.ovh.net SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport Max-Forwards: 70 From: asterisk sip:aster...@172.20.1.1;tag=as1545fb99 To: sip:sip.ovh.net Contact: sip:aster...@172.20.1.1 Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.8 Date: Wed, 08 Apr 2009 05:57:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '578ac87b06eaa6526aa313e130be3...@172.20.1.1' Method: OPTIONS [Apr 8 07:57:48] NOTICE[25949]: chan_sip.c:9490 sip_reg_timeout:-- Registration for '0033972112...@91.121.129.17' timed out, trying again (Attempt #1262) REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 91.121.129.17:5060: REGISTER sip:91.121.129.17 SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK40d5f950;rport Max-Forwards: 70 From: sip:0033972112...@91.121.129.17;tag=as02687bc2 To: sip:0033972112...@91.121.129.17 Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51 CSeq: 1466 REGISTER User-Agent: Asterisk PBX 1.6.0.8 Authorization: Digest username=0033972112355, realm=sip.ovh.net, algorithm=MD5, uri=sip:91.121.129.17, nonce=0019c92d503f745637b43af4264a11db, response=04e848af655d00e03d032d9a1c2fae09, opaque=001934772ef6ed5 Expires: 120 Contact: sip:0033972112...@172.20.1.1 Event: registration Content-Length: 0
Re: [asterisk-users] Using multiple 'peer' identities on one phone with 1.4
7 apr 2009 kl. 18.26 skrev Florian Hackenberger: On Tuesday 07 April 2009, Olle E. Johansson wrote: I don't see any problems there. YOu still have devices with states, as you would have with authentication. Of course, it still depends on your configuration. But authentication should not affect states. Ok, thanks for that, I'll have a look at openSER. If you use the limitonpeer setting, all states for both the user and the peer part of a friend will only be handled by the peer, which is the device watched for subscriptions. That worked like a charm, thanks! Good to hear. There was recently also an overhaul of the states for queues, with a patch to 1.4 that made it possible to build a stronger relationship between a queue member and a state object. Could you please point me to a bug report or an SVN revision? http://lists.digium.com/pipermail/asterisk-commits/2009-March/032220.html /O --- * Olle E. Johansson - o...@edvina.net * Asterisk Training http://edvina.net/training/ ___ -- 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 and Voice Recognition Sphinx
Hi all, someone has used the voice recognition software named Sphinx??? Can he tell me how to use and its performance??? Thanks Marco ___ -- 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] Siemens Gigaset Phones get mute function.
Hi, I know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. One of the biggest bug bears has been no mute function on the handset. When I woke up this morning, the handset told me there was a firmware update. I updated and then visited the web site to find out what had been fixed (quite a lot of new features have been added): http://gigaset.com/shc/0,1935,hq_en_0_152411_rArNrNrNrN_variation%253A-5_pageType%253Adownloads_imagePos%253A0,00.html#content Gigaset C470 IP / C475 IP / S675 IP / S685 IP Firmware update 04/2009 Download version: 02184 New features: * E-mail viewer (with C47H, S45, S67H, S68H handsets) * Mute function. Turn off the handset's microphone during an external call with the left display key. * Send and receive SMS messages via VoIP* * VoIP: If the telephone cannot establish a VoIP connection, it automatically dials via the fixed line network (auto-fallback to PSTN). * VoIP: Call transfer via R key * VoIP: An incoming call indicated in parallel at different VoIP devices (parallel ringing) will not be stored in the Missed calls list if the call was accepted at one of the devices.* * Online directory: display the postal codes in search results.* * Online directory: when starting a new search the cities used in the last searches will be displayed. * Fixed line access codes can be stored in the phone. * Some languages will be loaded on to the base via the internet, depending on the language set on the handset. * Extended RTP port range (1024-55000) * DHCP Option 114 implemented.* * DHCP Option 120 implemented.* * Web Configurator: option to specify whether the area code is dialled as well. * Web Configurator: display RTP port range * Web Configurator: new languages - Arabic and Russian * Web Configurator: enhanced PIN protection - warning if the default pin () has not been changed. Improvements: * The automatic search function for firmware updates is enabled even if the internet connection is temporarily interrupted during the night. * The indication Anonymous call activated will no longer be displayed in the idle mode of the handset. * The country code is synchronised between base station and handset. ___ -- 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] Voicemail and odbc storage (mysql)
Hello, Using odbc voicemail and mysql, i have a problem. After 12 seconds recording, asterisk stop recording and hangup. I have changed the settings in voicemail.conf to allow 180 seconds but,... Any hint? Olivier ___ -- 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] Voicemail and odbc storage (mysql)
On 8 Apr 2009, at 08:57, hh174 wrote: Using odbc voicemail and mysql, i have a problem. After 12 seconds recording, asterisk stop recording and hangup. I have changed the settings in voicemail.conf to allow 180 seconds but,... Any hint? Does it do it if you don't use odbc voicemail and mysql? If it does then its probably some gateway not liking the lack of two-way RTP. There is a variable to make it transmit silence. 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] Siemens Gigaset Phones get mute function.
On Wed, Apr 8, 2009 at 9:05 AM, Alan Lord (News) alansli...@gmail.com wrote: Hi, I know this is a little OT but there are many Asterisk users of the excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is probably newsworthy for them. Oh, yes! This is the greatest news since sliced bread! * VoIP: An incoming call indicated in parallel at different VoIP devices (parallel ringing) will not be stored in the Missed calls list if the call was accepted at one of the devices.* This was a good idea, too. Thanks for the post Alan, I saw the update notice on the phoine but didn't go for it until I ready this! r ___ -- 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] Siemens Gigaset Phones get mute function.
2009/4/8 Alan Lord (News) alansli...@gmail.com * DHCP Option 114 implemented.* * DHCP Option 120 implemented.* http://lists.digium.com/mailman/listinfo/asterisk-users What does it imply ? Provisionning from DHCP server ? ___ -- 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] Sangoma and BT single lines
2009/4/6 Ed W li...@wildgooses.com: Hi, got a Sangoma A200 with a bunch of extension cards and having real problems getting it to deal with a normal single BT line The A200 is a great card, and we use it quite a lot in the UK. Mostly we use the A200D for the echo cancellation. Symptoms are that incoming calls are fine. Outgoing calls ring the far end, BUT asterisk never sees that the call is answered (ie no message in the logs files saying so), as a result the remove end can hear the PBX side talking, but there is no audio back from the remote side to us. When we hangup the log files show messages thave suggest it thinks the line is still ringing I think someone else already said that Asterisk sends the number using DTMF, and then just opens up the audio channel - There is no answer detection involved beyond the user hearing the ringtone stop and the callee talking :) Comparing with another line which works fine (this is a BT multi-line system with what they call PBX signalling on it) I see that as soon as the remote end answers then asterisk gets a log message stating the same and audio is fine on this line We have found that using Residential settings as a starting point, and then asking for Disconnect clear time to be set to 800ms is all that is needed. That one setting allows the hangup to be detected reliably. We do also use the dialtone detection of Asterisk to be sure we're dialling on a line that is ready to take a call. Have now spent nearly 4 months trying to get the signalling sorted on this line. Most recently we requested dual signalling on the line - the end result is now that outbound calls work and asterisk reports that the phone answers, however, when you hangup the call then asterisk obviously gets a bunch of extra line reversals and things there is an immediate incoming call on the back of that outgoing call... I would turn off any line reversal detection - We found it horribly unreliable on BT's lines. Please - any suggestions on how to configure a Sangoma card for use with a normal BT single line? Fundamentally, BT suffer from backward compatability with the GPO syndrome, which basically means they do a load of stuff differently from the rest of the world. - Only the caller can ever hang up a call involving an analogue party. - Many line timings such as DCT are different and so forth Hope this is helpful... 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] [asterisk-dev] Grandstream blind transfer issue
Hi All, Thanks for your reply. I got this refer message in asterisk. but there is not any active channel of blind transfer. -- REFER sip:1...@192.168.1.25 sip%3a1...@192.168.1.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0 From: sip:7...@192.168.1.30:5060;transport=udp;tag=3699e1bcbed17687 To: 1101 sip:1...@192.168.1.25 sip%3a1...@192.168.1.25;tag=as32ed6c48 Contact: sip:7...@192.168.1.30:5060;transport=udp Supported: replaces, path Refer-To: sip:1631...@192.168.1.25 sip%3a1631...@192.168.1.25 Referred-By: sip:7...@192.168.1.25 sip%3a7...@192.168.1.25 Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 CSeq: 34526 REFER User-Agent: Grandstream BT200 1.1.6.46 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 - --- (14 headers 0 lines) --- Call 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 1631...@outgoing by 7...@192.168.1.25 localhost*CLI --- Transmitting (NAT) to 192.168.1.30:5060 --- SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.1.30:5060 ;branch=z9hG4bK5880efa5cca586b0;received=192.168.1.30 From: sip:7...@192.168.1.30:5060;transport=udp;tag=3699e1bcbed17687 To: 1101 sip:1...@192.168.1.25 sip%3a1...@192.168.1.25;tag=as32ed6c48 Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 CSeq: 34526 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:1...@192.168.1.25 sip%3a1...@192.168.1.25 Content-Length: 0 Is there any options we need to enable in asterisk or grandstream phone? I have already user transfer option 'Tt' in dialplan of this. Please provide me some help. Thanks in advance!! Thanks, Max Alex Voip Developer On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Max Alex wrote: Hi All, I have working asterisk version 1.4.24. I have a blind transfer issue with grandstream bt200. Does it work with other phones? That means is it a Grandstream isue or a general issue? I have updated the latest firmware to the phone. The phone is sending the *refer* to asterisk but asterisk is not able to connect with the another call Why? some log messages would help us helping you. that i have checked in sip debug. I am using transfer button of the grandstream phone. Can anybody provide help for this issue? Please ask again on the user mailing lists and provide some log messages Thanks in advance!! Thanks, Max Alex Voip Developer ___ -- 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-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ -- 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] Grandstream blind transfer issue
Hi I have used the transfer operation this way. When i got a call on grandstream phone, i will receive it and press transfer button and enter transfer number and press send button. My call is disconnected but no call transfer from asterisk. Please advice me!! Thanks, Max Alex Voip Developer On Tue, Apr 7, 2009 at 11:12 PM, Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote: On Tue, 7 Apr 2009, Max Alex wrote: Hi All, I have working asterisk version 1.4.24. I have a blind transfer issue with grandstream bt200. I have updated the latest firmware to the phone. The phone is sending the *refer* to asterisk but asterisk is not able to connect with the another call that i have checked in sip debug. I am using transfer button of the grandstream phone. Can anybody provide help for this issue? Thanks in advance!! How are you doing the entire transfer operation? For blind transfers, I do: Push Transfer (caller is now on hold, you get a new dial-tone) dial extension and push SEND At this point, called phone rings and caller is immediately taken off hold and transfered to the new ringing phone... you can hang up at that point. Don't use the 'flash' key. I have many BT200's and GXP280's out there - this seems to work for them without any issues. Asterisk 1.2 though. Gordon ___ -- 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] Asterisk Trunk billing
Hello, I have a problem with Asterisk trunk billing. I have bought some number of trunks from a VoIP provider with his own rates. I am planning to sell some of these trunks to my clients with my own rates. The problem is: how to process this trunk, Can I process it as a normal SIP/IAX client (if yes how) and apply my billing rates to it. 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] Hacked
On Tuesday 07 April 2009 11:28:52 Tilghman Lesher wrote: The recent vulnerability had nothing to do with this, but with the ability of an attacker to scan a SIP server for legitimate usernames and passwords. This, by the way, merely took advantage of the SIP protocol, as written. Normally, SIP allows you to differentiate between invalid usernames (404) and invalid passwords (403). What we closed in the recent vulnerability patch was to allow administrators to send back 403, regardless of whether the username existed or not. By the way, I am VASTLY oversimplifying the return codes here for the sake of clarity. The actual return code is based upon a number of factors, but it is modeled to return the same responses as would a bad password with a legitimate user account (thus making it impossible, externally, to tell the difference between a legitimate user account and a non-existent user account). -- Tilghman ___ -- 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] app_backticks and 1.6
On Wednesday 08 April 2009 00:08:23 Olivier wrote: I've updated http://www.voip-info.org/wiki/view/Asterisk+func+shell with an example to test file existence. Why not just use the STAT() function, in that case? -- Tilghman ___ -- 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 Trunk billing
On 8 Apr 2009, at 14:13, abdelkader wrote: I have a problem with Asterisk trunk billing. I have bought some number of trunks from a VoIP provider with his own rates. I am planning to sell some of these trunks to my clients with my own rates. The problem is: how to process this trunk, Can I process it as a normal SIP/IAX client (if yes how) and apply my billing rates to it. You need to write a billing application. Or buy one. ___ -- 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] Sangoma and BT single lines
Steve Davies wrote: 2009/4/6 Ed W li...@wildgooses.com: We have found that using Residential settings as a starting point,and then asking for Disconnect clear time to be set to 800ms is all that is needed. That one setting allows the hangup to be detected reliably. We do also use the dialtone detection of Asterisk to be sure we're dialling on a line that is ready to take a call. When was this added to Asterisk?? For years now, outbound dialing begun WITHOUT detecting dial tone, requiring multiple w to be inserted in the dial string. Dial tone detection was/is long overdue. Anyone know when this was added? Wading through cryptic change logs makes no mention of addition of this feature. This should have been put in bold red letters! Also Sangoma provides really great support for their cards. If all else fails, consider contacting Sangoma John Novack -- Dog is my co-pilot ___ -- 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] dahdi_dummy: Unable to register DAHDI rtc driver
David Backeberg wrote: Hello there: I think I have a silly kernel configuration problem. I'm running: * vanilla 2.6.27.10 kernel built from source * dahdi-2.1.0.4 built from source So far so good, dahdi module loads just fine: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.4 when I try to: hal04 dahdi # modprobe dahdi_dummy FATAL: Error inserting dahdi_dummy (/lib/modules/2.6.27.10/dahdi/dahdi_dummy.ko): Input/output error kernel messages gives me: dahdi_dummy: Unable to register DAHDI rtc driver I'm probably doing something silly here. Then I was curious, so I backed up to a 2.6.25.9 kernel I already had, and dahdi_dummy loaded just fine: dahdi_dummy: RTC rate is 1024 Does anybody know whether: * something changed in mainline kernel that breaks dahdi * there was a new kernel parameter that I should have set differently? There is a kernel config parameter called CONFIG_HPET_EMULATE_RTC, that if defined in the kernel config, will cause the behavior that you're seeing. I have not looked to see at which version that option came in. However, I think support for RTC in dahdi_dummy is best dropped. There is a patch on mantis (http://bugs.digium.com/view.php?id=13930) against dahdi_dummy that allows it to provide accurate timing with standard kernel timers that you may want to try. Cheers, -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Asterisk Trunk billing
Hi, You can start by looking here : http://www.voip-info.org/wiki/view/Asterisk+billing Jimmy -Original Message- From: abdelkader2...@gmail.com Sent: Wed, 8 Apr 2009 15:13:21 +0200 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Trunk billing Hello, I have a problem with Asterisk trunk billing. I have bought some number of trunks from a VoIP provider with his own rates. I am planning to sell some of these trunks to my clients with my own rates. The problem is: how to process this trunk, Can I process it as a normal SIP/IAX client (if yes how) and apply my billing rates to it. 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] AEL2, BASE64_DECODE and hexadecimal
On Tuesday 07 April 2009 23:38:08 Olivier wrote: 2009/4/7 Mark Michelson mmichel...@digium.com Philipp Kempgen wrote: BTW (developer's question) is there a reason why SendText() resp. sendtext_exec() refuses to send zero-length data? I can't point to any specific reason. I assume that whoever wrote the application probably thought that attempting to send zero-length data was pointless and that if no data were passed to the application, it likely was due to an error by the user. The phone I'm working on (Thomson ST2030) would display in slow blinking, inversed letters (white on black) any text received in SIP MESSAGE. Display duration is unlimited. To erase an old message, you must send a single carriage return (or maybe an empty string). I'm wondering how many phones behave like this ? Maybe, sendtext should then be refactored to accommodate this. What does the phone do when you send a single space? -- Tilghman ___ -- 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] __ast_read: ast_read() called with no recorded file descriptor
All, Im having a problem with ReceiveFax where its generating a ton of these messages the entire time the receivefax app is running receiving my fax. [Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. Im running on Centos 5.2 with all patches. asterisk-1.6.0.9 asterisk-addons-1.6.0.1 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 spandsp-0.0.5pre4 The receivefax app works perfectly, ie i am able to receive the faxes, and what not, but these messages are filling up my logs. Any ideas what is causing them. I know i saw a message like 2-3 weeks ago about it, but that guy was having e1 problems as well. This is a pure sip environment at the moment. Any pointers would be appreciated. ___ -- 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] __ast_read: ast_read() called with no recorded file descriptor
Greg Kennedy wrote: All, Im having a problem with ReceiveFax where its generating a ton of these messages the entire time the receivefax app is running receiving my fax. [Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. Im running on Centos 5.2 with all patches. asterisk-1.6.0.9 asterisk-addons-1.6.0.1 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 spandsp-0.0.5pre4 The receivefax app works perfectly, ie i am able to receive the faxes, and what not, but these messages are filling up my logs. Any ideas what is causing them. I know i saw a message like 2-3 weeks ago about it, but that guy was having e1 problems as well. This is a pure sip environment at the moment. Any pointers would be appreciated. Please see the following bug reports: http://bugs.digium.com/view.php?id=14723 (About the error message) http://bugs.digium.com/view.php?id=14769 (About Fax stuff) The short answer is that it appears there are many places that call ast_read() when they probably shouldn't. The thing is, the error message is what's new, not the other behavior. In other words, there aren't any new problems, just a new error message that points to problems that have been around a long time, most of which probably aren't that big a deal to begin with. Mark Michelson ___ -- 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-dev] Grandstream blind transfer issue
Haven't you read my email? 1. Wrong list 2. Missing log entries (set debug 4, set verbose 4) klaus Max Alex schrieb: Hi All, Thanks for your reply. I got this refer message in asterisk. but there is not any active channel of blind transfer. -- REFER sip:1...@192.168.1.25 mailto:sip%3a1...@192.168.1.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0 From: sip:7...@192.168.1.30:5060;transport=udp;tag=3699e1bcbed17687 To: 1101 sip:1...@192.168.1.25 mailto:sip%3a1...@192.168.1.25;tag=as32ed6c48 Contact: sip:7...@192.168.1.30:5060;transport=udp Supported: replaces, path Refer-To: sip:1631...@192.168.1.25 mailto:sip%3a1631...@192.168.1.25 Referred-By: sip:7...@192.168.1.25 mailto:sip%3a7...@192.168.1.25 Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 mailto:4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 CSeq: 34526 REFER User-Agent: Grandstream BT200 1.1.6.46 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 - --- (14 headers 0 lines) --- Call 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 mailto:4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 1631...@outgoing by 7...@192.168.1.25 mailto:7...@192.168.1.25 localhost*CLI --- Transmitting (NAT) to 192.168.1.30:5060 http://192.168.1.30:5060 --- SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0;received=192.168.1.30 From: sip:7...@192.168.1.30:5060;transport=udp;tag=3699e1bcbed17687 To: 1101 sip:1...@192.168.1.25 mailto:sip%3a1...@192.168.1.25;tag=as32ed6c48 Call-ID: 4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 mailto:4d6a024a07f2b0f904a3cfe26360e...@192.168.1.25 CSeq: 34526 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:1...@192.168.1.25 mailto:sip%3a1...@192.168.1.25 Content-Length: 0 Is there any options we need to enable in asterisk or grandstream phone? I have already user transfer option 'Tt' in dialplan of this. Please provide me some help. Thanks in advance!! Thanks, Max Alex Voip Developer On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at wrote: Max Alex wrote: Hi All, I have working asterisk version 1.4.24. I have a blind transfer issue with grandstream bt200. Does it work with other phones? That means is it a Grandstream isue or a general issue? I have updated the latest firmware to the phone. The phone is sending the *refer* to asterisk but asterisk is not able to connect with the another call Why? some log messages would help us helping you. that i have checked in sip debug. I am using transfer button of the grandstream phone. Can anybody provide help for this issue? Please ask again on the user mailing lists and provide some log messages Thanks in advance!! Thanks, Max Alex Voip Developer ___ -- 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-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ -- 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] Sangoma and BT single lines
sangoma support are amazing, they've solved nearly all the problems i've experienced with PRI, except for one which turned out to be a bug in SWIX (some rubbish windows based voip pbx, full of bugs and generally crap!). there also quite happy to log in to your systems and have a look themselves if you want them to, or if it's a particularly mind boggling problem. 2009/4/8 John Novack jnov...@stromberg-carlson.org Steve Davies wrote: 2009/4/6 Ed W li...@wildgooses.com: We have found that using Residential settings as a starting point,and then asking for Disconnect clear time to be set to 800ms is all that is needed. That one setting allows the hangup to be detected reliably. We do also use the dialtone detection of Asterisk to be sure we're dialling on a line that is ready to take a call. When was this added to Asterisk?? For years now, outbound dialing begun WITHOUT detecting dial tone, requiring multiple w to be inserted in the dial string. Dial tone detection was/is long overdue. Anyone know when this was added? Wading through cryptic change logs makes no mention of addition of this feature. This should have been put in bold red letters! Also Sangoma provides really great support for their cards. If all else fails, consider contacting Sangoma John Novack -- Dog is my co-pilot ___ -- 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] Best Practice Advice?
Also, FC10 is out. You should probably grab that first. Unless you are a strong Linux Guru, I would never recommend a Fedora release for a production system. I have FC9 here and FC10. It took me months to eliminate the bugs from FC9, and I still haven't gotten FC10 to install on the machine I got it for (three months now). Fedora is cutting edge and puts out a new release probably every six months with less than usual regard for consistency or stability. I don't know of anything Asterisk that requires this level of cutting edge technology. While all the bugs I fought in FC9 are gone, they have been replaced by a whole new spate of (some still unidentified) bugs. Centos is a much more appropriate distro for production work. Nothing goes into it until it is known to be rock solid, and update occur much more slowly. It wouldn't be too far off base to say that Fedora users are the beta testers for Centos--not explicitly in terms of versions, but certainly in terms of features and code base. I'm sure there are other good (maybe even better) distros for Asterisk, I'm not familiar with all of them. Fedora is really at home with someone who is running a personal web server or media computer or something for a hobby and likes to have the latest of everything and wants to (or at least is willing to) play with it, get it to work and help improve it. That isn't the recipe for running a business. Wilton ___ -- 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] Zopier Client
Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg ___ -- 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] Best Practice Advice?
Centos is a much more appropriate distro for production work. Nothing goes into it until it is known to be rock solid, and update occur much more slowly. Yes, and that also means newer glib etc can be needed sometimes which are not YET avail on centos, however if you are not a yum freak and prefer to control builds, its perfect. Been using centos since ever, and never got any problems, never.. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wilton Helm Sent: April-08-09 12:40 PM To: Asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Best Practice Advice? Also, FC10 is out. You should probably grab that first. Unless you are a strong Linux Guru, I would never recommend a Fedora release for a production system. I have FC9 here and FC10. It took me months to eliminate the bugs from FC9, and I still haven't gotten FC10 to install on the machine I got it for (three months now). Fedora is cutting edge and puts out a new release probably every six months with less than usual regard for consistency or stability. I don't know of anything Asterisk that requires this level of cutting edge technology. While all the bugs I fought in FC9 are gone, they have been replaced by a whole new spate of (some still unidentified) bugs. Centos is a much more appropriate distro for production work. Nothing goes into it until it is known to be rock solid, and update occur much more slowly. It wouldn't be too far off base to say that Fedora users are the beta testers for Centos--not explicitly in terms of versions, but certainly in terms of features and code base. I'm sure there are other good (maybe even better) distros for Asterisk, I'm not familiar with all of them. Fedora is really at home with someone who is running a personal web server or media computer or something for a hobby and likes to have the latest of everything and wants to (or at least is willing to) play with it, get it to work and help improve it. That isn't the recipe for running a business. Wilton ___ -- 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] Zopier Client
Gregory Malsack wrote: Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg I've been using it on my notebook. I've been happy with it but I'm not a heavy user. The biggest reason I purchased a few copies of it is that I need to have several different sip and iax2 connections for testing purposes. -- Darren Wiebe dar...@aleph-com.net Aleph Communications www.aleph-com.net ___ -- 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] Call Pickup Works w/Linksys ATA, not with Cisco 7940G
On Tue, 7 Apr 2009, George Pajari wrote: I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G phones. Unit obtains SIP trunking from an ITSP (server has no PCI boards). *8 Call Pickup works fine from any of the phones connected using the Linksys SPA2102. *8 Call Pickup does not work from the Cisco 7940G phones (chan_sip.c:13977 handle_request_invite: Nothing to pick up for 000d6556-eeb3001c-76b88543-7f51d...@192.168.0.211) Seems someone else had the same problem back in 2004 and got no answer. http://lists.digium.com/pipermail/asterisk-users/2004-April/036869.html Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- 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] Zopier Client
On Wed, 8 Apr 2009, Gregory Malsack wrote: Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? I've used it - the business edition supports more than 2 accounts. I think that's the main difference. (Both now support SIP and IAX) I liked it when it was IDEFISK, but sort of liking it less now that it's Zoiper - however I'm using it (or trying to use it) under Linux. If you're using it under Windows then it's probably fine. The issues I had under Linux were to do with library incompatabilities - and sound system incompatabilities! They had also built it under a bleeding edge version of Ubuntu and I was trying to run it under Debian stable (which at the time was Etch). I've more or less given up on it under Linux and have installed my old idefisk under Debian Lenny on my notebooks. I had email from them hinting of things to come - video support and so on. Personally all I want is an IAX command-line soft-phone and I'll write one myself if I ever get the time. Gordon ___ -- 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] Zopier Client
On 4/8/2009 1:19 PM, Gregory Malsack wrote: Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg I am not a very heavy user of it either, but I'm a semi-regular user, and I like it a lot. It's the most stable and usable IAX2/SIP soft-phone I have used, and I've used at least a dozen of them before finally settling on Zoiper, and then Zoiper-Biz. I don't use some of the fancier features, but what I do use, always works as expected. Call quality is very good too. ___ -- 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] Zopier Client
Gordon Henderson wrote: On Wed, 8 Apr 2009, Gregory Malsack wrote: Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? I've used it - the business edition supports more than 2 accounts. I think that's the main difference. (Both now support SIP and IAX) I liked it when it was IDEFISK, but sort of liking it less now that it's Zoiper - however I'm using it (or trying to use it) under Linux. If you're using it under Windows then it's probably fine. The issues I had under Linux were to do with library incompatabilities - and sound system incompatabilities! They had also built it under a bleeding edge version of Ubuntu and I was trying to run it under Debian stable (which at the time was Etch). I've more or less given up on it under Linux and have installed my old idefisk under Debian Lenny on my notebooks. I know about that problem and we are trying to tackle it, expect some new builds very soon. (including .deb and .rpms). I had email from them hinting of things to come - video support and so on. Personally all I want is an IAX command-line soft-phone and I'll write one myself if I ever get the time. Zoiper should do command line, although it will still run the gui i guess :( Gordon ___ -- 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] Zopier Client
Gregory Malsack wrote: Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? I have some experience with it, as i'm on the team producing it :) Can't give you unbiased comments though, so i guess its better if you try it yourself, just mail sa...@zoiper.com tell them your name and ask for the free biz copy i promised you :) Zoa Thanks, Greg ___ -- 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] Best Practice Advice?
Wilton Helm schrieb: Also, FC10 is out. You should probably grab that first. And by the way: Debian 5 Lenny is out. http://www.debian.org/ Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Siemens Gigaset Phones get mute function.
2009/4/8 Olivier oza-4...@myamail.com 2009/4/8 Alan Lord (News) alansli...@gmail.com * DHCP Option 114 implemented.* * DHCP Option 120 implemented.* http://lists.digium.com/mailman/listinfo/asterisk-users What does it imply ? Provisionning from DHCP server ? 114 is for passing a URL to be displayed after boot 120 is for passing SIP servers (RFC3361) So, I guess that would be a yes... d ___ -- 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] Best Practice Advice?
On Wed, 8 Apr 2009, Philipp Kempgen wrote: Wilton Helm schrieb: Also, FC10 is out. You should probably grab that first. And by the way: Debian 5 Lenny is out. http://www.debian.org/ I've been a Debian user since more or less the begining (of Debian that is - there was sls before that!) I tend to use it as a base for the basic OS and compiler and compile most other big userland stuff though. So I compile up asterisk from scratch (along with the kernel, apache, php, mysql, etc.) but leave the basic system stuff alone. (This stems from a time when I was maintaning many different *nixes, but wanted a common set of tools over them) I've never really had issues compiling up asterisk, etc. for all the versions of Debian I've used with it - Sarge, Etch and now Lenny all work very well. And Debian has been stable if nothing else. I have a Debian box with over 3 years of uptime (it's not running asterisk though) and some of my PBXs which run my own install based on Debian have 6 months uptime So if you want a stable platform, get Debian! Gordon ___ -- 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] Perl AGI
Hi all,
I have the below peace of my AGI script...the problem here is that I cannot
fetch the extension value to inside the script and assign it to another
variable...I highlighted it in red
#!/usr/bin/perl
#use DBD::mysql;
use DBI;
use DBD::mysql;
use Asterisk::AGI;
#To read asterisk variable values.
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-verbose(AGI Environment Dump:, 3);
my $micho = $input{9};
$AGI-verbose(my dialed no is :$micho);
foreach my $i (sort keys %input) {
$AGI-verbose( -- $i = $input{$i}, 3);
}
##
#To get the asterisk dial no whihc is 112 in our case
my $no=$AGI-get_variable ('extnum');
*my $dest=$AGI-get_variable ('extension');*
$AGI-verbose(my dialed no is :$no);
$AGI-verbose(my dialed no is :$dest);
When the script run I got: dial.pl: my dialed no is :
Can you please help me to fix this issue?
Regards
___
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Re: [asterisk-users] Hacked
Here's what fail2ban service caught The IP 89.111.184.221 has just been banned by Fail2Ban after 80 attempts against ASTERISK. On Wed, Apr 8, 2009 at 7:01 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Tuesday 07 April 2009 11:28:52 Tilghman Lesher wrote: The recent vulnerability had nothing to do with this, but with the ability of an attacker to scan a SIP server for legitimate usernames and passwords. This, by the way, merely took advantage of the SIP protocol, as written. Normally, SIP allows you to differentiate between invalid usernames (404) and invalid passwords (403). What we closed in the recent vulnerability patch was to allow administrators to send back 403, regardless of whether the username existed or not. By the way, I am VASTLY oversimplifying the return codes here for the sake of clarity. The actual return code is based upon a number of factors, but it is modeled to return the same responses as would a bad password with a legitimate user account (thus making it impossible, externally, to tell the difference between a legitimate user account and a non-existent user account). -- Tilghman ___ -- 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] Perl AGI
This is at least correct on my setup
$dest = $input{dnid}
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha
Sent: Wednesday, April 08, 2009 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Perl AGI
Hi all,
I have the below peace of my AGI script...the problem here is that I cannot
fetch the extension value to inside the script and assign it to another
variable...I highlighted it in red
#!/usr/bin/perl
#use DBD::mysql;
use DBI;
use DBD::mysql;
use Asterisk::AGI;
#To read asterisk variable values.
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-verbose(AGI Environment Dump:, 3);
my $micho = $input{9};
$AGI-verbose(my dialed no is :$micho);
foreach my $i (sort keys %input) {
$AGI-verbose( -- $i = $input{$i}, 3);
}
##
#To get the asterisk dial no whihc is 112 in our case
my $no=$AGI-get_variable ('extnum');
my $dest=$AGI-get_variable ('extension');
$AGI-verbose(my dialed no is :$no);
$AGI-verbose(my dialed no is :$dest);
When the script run I got: dial.pl: my dialed no is :
Can you please help me to fix this issue?
Regards
___
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Re: [asterisk-users] Perl AGI
It works d\Danny...Thanks a lot for your help
Regards
On Thu, Apr 9, 2009 at 12:37 AM, Danny Nicholas da...@debsinc.com wrote:
This is at least correct on my setup
$dest = $input{dnid}
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha
*Sent:* Wednesday, April 08, 2009 4:28 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Perl AGI
Hi all,
I have the below peace of my AGI script...the problem here is that I cannot
fetch the extension value to inside the script and assign it to another
variable...I highlighted it in red
#!/usr/bin/perl
#use DBD::mysql;
use DBI;
use DBD::mysql;
use Asterisk::AGI;
#To read asterisk variable values.
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-verbose(AGI Environment Dump:, 3);
my $micho = $input{9};
$AGI-verbose(my dialed no is :$micho);
foreach my $i (sort keys %input) {
$AGI-verbose( -- $i = $input{$i}, 3);
}
##
#To get the asterisk dial no whihc is 112 in our case
my $no=$AGI-get_variable ('extnum');
*my $dest=$AGI-get_variable ('extension');*
$AGI-verbose(my dialed no is :$no);
$AGI-verbose(my dialed no is :$dest);
When the script run I got: dial.pl: my dialed no is :
Can you please help me to fix this issue?
Regards
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Re: [asterisk-users] Hacked
Nice, share the knowledge and send the fail2ban rule ;) ill post mine's From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaswinder Singh Sent: April-08-09 5:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hacked Here's what fail2ban service caught The IP 89.111.184.221 has just been banned by Fail2Ban after 80 attempts against ASTERISK. On Wed, Apr 8, 2009 at 7:01 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Tuesday 07 April 2009 11:28:52 Tilghman Lesher wrote: The recent vulnerability had nothing to do with this, but with the ability of an attacker to scan a SIP server for legitimate usernames and passwords. This, by the way, merely took advantage of the SIP protocol, as written. Normally, SIP allows you to differentiate between invalid usernames (404) and invalid passwords (403). What we closed in the recent vulnerability patch was to allow administrators to send back 403, regardless of whether the username existed or not. By the way, I am VASTLY oversimplifying the return codes here for the sake of clarity. The actual return code is based upon a number of factors, but it is modeled to return the same responses as would a bad password with a legitimate user account (thus making it impossible, externally, to tell the difference between a legitimate user account and a non-existent user account). -- Tilghman ___ -- 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] Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA
Hi, I have installed Elastix 1.5.2 (Barranquilla, Colombia (TELCO: METROTEL)) with a TE220P (2xE1) and TDM2400P (16FXS), openr2 is included in 1.5.2 version. The outcoming calls are ok, but with incoming call i have an error: ERROR*[*14972*]* chan_dahdi.c: Chan 2 - Protocol error. Reason = Multi Frequency Cycle Timeout, R2 State = Seize ACK Transmitted, MF state = Category Request Transmitted, MF Group = Backward Group A, CAS = 0x00 DNIS = 310, ANI = , MF = 0x20 I tried with all protocol variants availables, because seems thats the cause, but I still have the problem. elastix*CLI mfcr2 show variant Variant Code Country ARArgentina BR Brazil CNChina CZ Czech Republic CO Colombia EC Ecuador ITUInternational Telecommunication Union MX Mexico PH Philippines VEVenezuela elastix*CLI The following link has the content of files: chan_dahdi.conf, system.conf, and a tail of /var/log/asterisk/full http://pastebin.com/f3424b319 Is this really a variant protocol problem? Any suggest? Regards, GM ___ -- 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] Softphone question
I'm afraid I already know the answer because I've done a lot of searching, but does anyone know of a softphone that supports a central phone book and paging (like the sip autoanswer option of some hardphones) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.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] Softphone question
Xlite etc, counterpath.com have AA features, not sure about central phone book. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: April-08-09 10:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Softphone question I'm afraid I already know the answer because I've done a lot of searching, but does anyone know of a softphone that supports a central phone book and paging (like the sip autoanswer option of some hardphones) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.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 ___ -- 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 and WebIntegration
Hi Geraint, My apologies for the very very late reply. But, I wasn't able to make the incoming calls park in one extension and pick the call from there. The agents are quite comfortable with the setup we discussed, as calls they make will be made ring in their EyeBeam and then gets connected to the external number. The incoming call (which arrives) will be waiting in the second line. I was in the process of designing a training program in asterisk which asterisk will ask a set of questions to the agents, records the response from the agent and saves as gsm/wav file and then mix the question and answer. So was kind of tied up. Anyways, thank you so much for your help and suggestions. I really appreciate it. Keep posting and keep helping. Regards, Kurian Mathew Thayil. On Fri, Mar 13, 2009 at 4:58 PM, Geraint Lee gera...@gmail.com wrote: I reverse the inbound calls so they appear as outbound calls for agents, all of our calls are managed by the dialer i've written and integrate directly to our CRM, essentially asterisk is only providing the SIP/IAX functionality to me everything else is done via php... so... inbound call comes in and gets parked in a php script stores in database as an outbound call, agents screen then pops and checks the database for the CLI so we can try to guess who's calling us and opens up all of their details. php script that is parking the inbound call then dials the allocated agents extension and connects the call. also on the dial command i have used Dial(SIP/1234,,A(beep)) so that the agent hears a beep when they get a call. Hope this enlightens you a bit on handling inbounds in this situation :) Cheers 2009/3/12 Kurian Thayil kurianmtha...@gmail.com Hi Geriant, My apologies for the delay in reply. We won't be using php but Perl and there is an AGI module for perl Asterisk::AGI. I may be using Manager API for sending Hangup signal. Im planning to write a bash script which perl invokes when hangup button is pressed in the web interface. Bash script telnets and sends Hangup signal to the manager API. I am not yet able to acheive sending commands via bash script using telnet. But I am trying. One thing that's confusing me is if in future, incoming facility needs to be activated and since Auto Answer feature in EyeBeam/Twinkle is ON, don't you think that would be a problem? I think for that, the possible work around will be using 2 softphones, say EyeBeam and Xlite together in the same PC. Configuring one extension in EyeBeam to make outbound calls (with Auto Answer enabled) and configuring Xlite with an extension which receives inbound calls. Do you have any suggestion on that? Regards, Kurian Mathew Thayil. On Tue, Mar 10, 2009 at 7:32 PM, Geraint Lee gera...@gmail.com wrote: If you're using a php i'd take a look at phpagi - there are others around for various different languages too. our agents use twinkle with auto-answer, the only reason they need to look at twinkle is if they need to perform a transfer (that too will soon be done from the web browser), you can do pretty much anything with the asterisk manager (originate the call and hangup the call and a load of other useful stuff) Cheers 2009/3/10 Kurian Thayil kurianmtha...@gmail.com Hi Steve, That worked beautifully. Thank you so much. But one question though. Imagine if I keep a Hangup Button in the interface and it should terminate the call. Will that be possible? This scenario happens when the user gets connected to an invalid phone number where the user have to manually disconnect. I don't plan to confuse the user by asking them to use eyebeam to disconnect the call. If it could be integrated to the web interface they just have to stick on to that alone. Is there any way? Regards, Kurian Mathew Thayil. On Tue, Mar 10, 2009 at 4:51 PM, Steve Totaro stot...@first-notification.com wrote: On Tue, Mar 10, 2009 at 6:40 AM, Kurian Thayil kurianmtha...@gmail.com wrote: Hi All, Is there a way that I can include call dialing functionality in a webinterface. I have EyeBeam configured with a SIP user say 8440. Will I be able to design an inteface which agent can choose a number and the Dial without punching in the number in Eyebeam. I tried using the .call file. ie The agent can choose which number to dial from a web interface. Then, a .call file is created with the following informations. Channel: Zap/g2/9444204943 Context: inbound_support Extension: 8440 Priority: 0 Now, in the extensions.conf file, I mentioned the following under inbound_support context. [inbound_support] exten =8440,1,Dial(SIP/8440,55,tTo) exten =8440,2,Answer exten =8440,3,Hangup But, here the call gets connected only when the receiver end receives the call. When the receiver end picks up the phone, SIP/8440 rings. Is there any other way to implement this. I am not ready to use Vicidial (AstGUIClient) because the interface to be designed is too
Re: [asterisk-users] Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA
Sounds like a protocol variant issue. Is the telco supposed to send you ANI? You have 2 options, the first option is to try with the ITU variant, if that does not work, set the option mfcr2_skip_category=yes and see if that helps. Moy On Wed, Apr 8, 2009 at 6:06 PM, Giovanny Magallanes gmagalla...@gmail.com wrote: Hi, I have installed Elastix 1.5.2 (Barranquilla, Colombia (TELCO: METROTEL)) with a TE220P (2xE1) and TDM2400P (16FXS), openr2 is included in 1.5.2 version. The outcoming calls are ok, but with incoming call i have an error: ERROR[14972] chan_dahdi.c: Chan 2 - Protocol error. Reason = Multi Frequency Cycle Timeout, R2 State = Seize ACK Transmitted, MF state = Category Request Transmitted, MF Group = Backward Group A, CAS = 0x00 DNIS = 310, ANI = , MF = 0x20 I tried with all protocol variants availables, because seems thats the cause, but I still have the problem. elastix*CLI mfcr2 show variant Variant Code Country AR Argentina BR Brazil CN China CZ Czech Republic CO Colombia EC Ecuador ITU International Telecommunication Union MX Mexico PH Philippines VE Venezuela elastix*CLI The following link has the content of files: chan_dahdi.conf, system.conf, and a tail of /var/log/asterisk/full http://pastebin.com/f3424b319 Is this really a variant protocol problem? Any suggest? Regards, GM ___ -- 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 -- I do not agree with what you have to say, but I’ll defend to the death your right to say it. Voltaire ___ -- 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] agi no longer working with 1.4 svn 186229
On 4/04/2009 2:22 a.m., John covici wrote: The minute asterisk tries to execute an agi, it gets utils.c write error broken pipe and so hangs up the call. Anyone know what is going on? I am using kernel 2.6.27 with dahdi trunk if that makes a difference. thanks in advance for any ideas. Can you run the AGI from the Linux command line? Just press enter lots to get into it. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] agi no longer working with 1.4 svn 186229
If I revert back to my old version of asterisk, it works just fine. on Thursday 04/09/2009 Matt Riddell(li...@venturevoip.com) wrote On 4/04/2009 2:22 a.m., John covici wrote: The minute asterisk tries to execute an agi, it gets utils.c write error broken pipe and so hangs up the call. Anyone know what is going on? I am using kernel 2.6.27 with dahdi trunk if that makes a difference. thanks in advance for any ideas. Can you run the AGI from the Linux command line? Just press enter lots to get into it. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.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] OT - SIP MESSAGE, newline chars and formatting
On 7/04/2009 9:41 p.m., Olivier wrote: Hi, I'm using a SIP phone (Thomson ST2030) which is able to display text received though Asterisk's SendText() application. I'm using this to display from Asterisk Forwarded to 0123456789 whenever a user forwards his calls to another number or extension. Test is displayed with white letters on black background. What I can't do at the moment is erasing this Forwarded to 0123456789 text when user cancels previous forwarding. If I'm sending a string full of space chars, I've got an ugly string of black rectangles on LCD screen. Phone vendor says it can be done sending a single carriage return string to the phone (using usual SendText, I suppose) but either : A- I can't build correctly such single carriage return string, B- I can't send it (I shouldn't use SendText()), C- or I misunderstood vendor's advice. Have you tried with \r\n, \n, or even maybe br / and br -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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
