[asterisk-users] Fwd: FW: loadstar and ok one
Hi All, I have experiancing strenge issue with my production Asterisk system. I'm using asterisk vertion 1.4.28 installed cent OS 5. Issue decription. I have SIP trunk from local carrier to their hosted PBX( broadsoft). Out going calls over this trunk working fine and I can make a conversation with landlines and mobiles but incomming not working. I can see calls are hitting my IVR but no audio. This system worked till yesterday without any issue. Please find attached SIP traces from received from my carrier and what they are saying is they not receiving proper information on SIP 200 message. I'm attached my asterisk system traces also named asterisk SIP log. Please looking to this and provide me help ASAP. Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.36:5061;branch=z9hG4bKooi3r32663oonttllrzts6lio Call-ID: 63kothjzr4m4ks2jsimktm64ihrsrhlk@SoftX3000 From: sip:717747766@10.10.1.36;user=phone;tag=t4h4mk46-CC-24 To: sip:600@10.20.1.66;user=phone;tag=aprqtqg8kq3-mt4au3220 CSeq: 1 CANCEL --- Retransmitting #6 (NAT) to 10.8.55.194:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.55.194:5060;branch=z9hG4bKrjogsq209000rjk8s-3.1;received=10.8.55.194 From: sip:773208775@10.8.55.194;user=phone;tag=SD5j4u701-or643njn-CC-23 To: sip:600@10.94.0.45;user=phone;tag=as18171f50 Call-ID: SD5j4u701-8b8b9bbad4be07ce8971c1f3532859ab-ag220u0 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: sip:600@10.94.0.45 ontent-Type: application/sdp Content-Length: 304 v=0 o=root 29336 29336 IN IP4 10.94.0.45 s=session c=IN IP4 10.94.0.45 t=0 0 m=audio 10802 RTP/AVP 8 0 18 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- elastix*CLI --- SIP read from 194.78.20.125:1092 --- REGISTER sip:202.124.179.130 SIP/2.0 From: Tweco Wimsip:1103@202.124.179.130;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81 To: Tweco Wimsip:1103@202.124.179.130 Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b@10.32.0.127 CSeq: 1860 REGISTER Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbe-212749c0-7fc98295 Max-Forwards: 70 Supported: replaces,net2phone User-Agent: LGE LIP 6812 v1.1.31s SN/00405A181B6A Contact: sip:1103@194.78.20.125:1093 Expires: 60 Authorization: Digest username=1103,realm=asterisk,nonce=0dcbdd61,uri=sip:202.124.179.130,response=8ff1c4bfbd6ad4a9dc5db70640dfc431,algorithm=MD5 Content-Length: 0 - --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 194.78.20.125 : 1092 (NAT) --- Transmitting (NAT) to 194.78.20.125:1092 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbe-212749c0-7fc98295;received=194.78.20.125 From: Tweco Wimsip:1103@202.124.179.130;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81 To: Tweco Wimsip:1103@202.124.179.130 Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b@10.32.0.127 CSeq: 1860 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- Transmitting (NAT) to 194.78.20.125:1092 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbe-212749c0-7fc98295;received=194.78.20.125 From: Tweco Wimsip:1103@202.124.179.130;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81 To: Tweco Wimsip:1103@202.124.179.130;tag=as5c46c474 Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b@10.32.0.127 CSeq: 1860 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=3b3b16d9 Content-Length: 0 Scheduling destruction of SIP dialog '3131303300-aabb-5b49-0405a181b6a-0-3b@10.32.0.127' in 32000 ms (Method: REGISTER) elastix*CLI --- SIP read from 194.78.20.125:1092 --- REGISTER sip:202.124.179.130 SIP/2.0 From: Tweco Wimsip:1103@202.124.179.130;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81 To: Tweco Wimsip:1103@202.124.179.130 Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b@10.32.0.127 CSeq: 1861 REGISTER Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbf-21274a88-2be0b1a2 Max-Forwards: 70 Supported: replaces,net2phone User-Agent: LGE LIP 6812 v1.1.31s SN/00405A181B6A Contact: sip:1103@194.78.20.125:1093 Expires: 60 Authorization: Digest username=1103,realm=asterisk,nonce=3b3b16d9,uri=sip:202.124.179.130,response=1d39bc50536a47031997a5c40cd65402,algorithm=MD5 Content-Length: 0 - --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 194.78.20.125 : 1092 (NAT) --- Transmitting (NAT) to 194.78.20.125:1092 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbf-21274a88-2be0b1a2;received=194.78.20.125
[asterisk-users] IAX endpoints not Registering after upgrage from Asterisk ver 1.4.26.1
Dear All, I am experiance a issue with my IAX clients. I have upgradeed Asterisk to 1.4.28 After then IAX clients are not working and It's not registering even. Please help. Asterisk previous version - 1.4.26.1 ( for this worked fine) FreePBX version - freepbx-2.5.2 -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Install dahdi on Xen virtual console
Hi, We are trying compile dahdi on amazon vertual instance. When we are compiling dahdi we receieve following error. You do not appear to have the sources for the 2.6.21.7-2.fc8xen kernel installed. We are helpless on getting this 2.6.21.7-2 sources. Please help to get this compile. -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX devices not registering after upgrade to asterisk 1.4.29
Hi All, We have encountering issue that IAX enable voice gateways not registering with asterisk after upgrade from asterisk 1.4.18.1 - 1.4.29 Before that IAX works very well. If any one have similar issue and solution for that let me know. -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased -- _ -- 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] IAX devices not registering after upgrade to asterisk
Message: 18 Date: Tue, 23 Feb 2010 15:02:24 +0100 From: Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de Subject: Re: [asterisk-users] IAX devices not registering after upgrade to asterisk 1.4.29 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 4b83ee00.10686.84d...@klitzing.pool.informatik.rwth-aachen.de Content-Type: text/plain; charset=US-ASCII Hi philip, The Issue sorted. thanks for sharing the details. Regards, Vidura. Hi! We have encountering issue that IAX enable voice gateways not registering with asterisk after upgrade from asterisk 1.4.18.1 - 1.4.29 Before that IAX works very well. If any one havesimilarissue and solution for that let me know. Search or google for calltokenoptional. http://www.voip-info.org/wiki/view/Asterisk+config+iax.conf http://downloads.asterisk.org/pub/security/AST-2009-006.html Philipp -- _ -- 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] ISP -Asterisk - ATA -DIALUP
Hellow, * I have a problem with dial up signalling. currently I have configured asterisk server and E1 card to ISP. then other side I am having ATA to PC for connecting internet through DialUP connection. is it possible and please send me the procedure how I can do it ?? * ISP - Asterisk - ATA - DIALUP -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased ___ -- 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] Can Asterisk bridge between a SIP client and a Cisco Call
Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes your life easy. -- Thanks Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased == Message: 16 Date: Fri, 10 Apr 2009 00:06:50 -0600 From: Shocky shoc...@users.sourceforge.net Subject: [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call Manager server? To: asterisk-users@lists.digium.com Message-ID: 20090416.51201.shoc...@users.sourceforge.net Content-Type: text/plain; charset=us-ascii Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. Thanks, Shocky -- These are my opinions. Get your own. -- Message: 17 Date: Fri, 10 Apr 2009 10:07:38 +0300 From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] MeetMe not working - was before To: asterisk-users@lists.digium.com Message-ID: 20090410070738.gs3...@xorcom.com Content-Type: text/plain; charset=us-ascii On Thu, Apr 09, 2009 at 04:59:41PM -0400, John Rogers wrote: When I dial the extension of a meetme conference room, I get a message that states is not a valid conference. The meetme app was working before. I am getting this error on the CLI: app_meetme.c:800 build_conf: Unable to open pseudo device I have Asterisk 1.4.23.1 and zaptel-1.4.11 Elsewhere you mentioned you also have dahdi installed. What is the output of: ls /usr/include/dahdi I suspect Asterisk was built vs. dahdi whereas Zaptel was actually running. Actual tests: dahdi_test vs. zttest -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- Message: 18 Date: Fri, 10 Apr 2009 10:33:36 +0100 (BST) From: Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net Subject: Re: [asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call Manager server? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: pine.lnx.4.64.0904101032040.23...@unicorn.drogon.net Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Fri, 10 Apr 2009, Shocky wrote: Hi, This is probably outside what Asterisk is intended for, but I'm hoping it can help. I need to make and receive calls through a Cisco Call Manager server that I have no control over. I have to use a Cisco soft phone (Cisco IP Communicator), which only runs on Windows. But I'm on Linux. CCM is apparently capable of supporting SIP and H.323 interfaces, but they won't provide this option for me. Right now I'm using a VMWare XP guest to run the soft phone, but this is painful (especially with some VPN complications thrown in). I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I could set up Asterisk on my desktop machine to route calls between a SIP client such as Kphone or Ekiga and the CCM server. Would this be possible? I heard that one of the problems in interfacing with CCM over SCCP is the use of proprietary codecs. Would this be a problem in my case? If there's a chance it can be made to work, I'll give it a try. If I'd be wasting my time, please let me know. I've never looked at SCCP, but if it does work then you could use the console phone built into asterisk rather than IP plumb it into a soft-phone... So asterisk is essentially acting as an SCCP soft-phone itself. No GUI though
Re: [asterisk-users] asterisk-users Digest, Vol 48, Issue 4
Hi, Check sip.conf settings. disable TCP and TLS, or if there is any securify related parameters. Use UDP and test. Send us your feedback. Regards, Vidura B. Senadeera. -- Message: 4 Date: Tue, 1 Jul 2008 13:50:16 -0400 From: David Siegel [EMAIL PROTECTED] Subject: [asterisk-users] Broadvoice and Asterisk 1.6.0-beta9 To: 'asterisk-users@lists.digium.com' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I am still failing to get a Broadvoice SIP peer to work correctly with Asterisk 1.6.0-beta9. My setup works fine with Asterisk 1.2. After upgrading to 1.6, I getting a FORBIDEN error from Broadvoice when Asterisk attempts to connect. I'm wondering if anyone has a working SIP connect to Broadvoice with the latest release of Asterisk, and if they do, could they share their configuration setup with me. Thank you! -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080701/902c4d59/attachment-0001.htm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ip phone suggestion for Asia?
Hi, Try atcom. www.atcom.com.cn We have tested atcom and its quality also good. they are using infeneon chipset. its support asterisk, sip, iax as well.decent look. cost effetive. still they have basic ip phone modles. starting from next year they will release new modles. Regards, vidura. -- Message: 1 Date: Fri, 21 Dec 2007 12:04:57 +0100 From: Fredrik S?derlund [EMAIL PROTECTED] Subject: Re: [asterisk-users] ip phone suggestion for Asia? To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Check out yntx www.yntx.com fear prices and recides in Asia and iss it sip on asteriks they will do ! try to buy one to trye it out before buying fore hole company.. /MVH Fille ___ --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] Didnt get a frame from Channel and call gets
Hi, Let us know more information about your setup. Hardware/software details details such as. server configuration PSTN cards you are using?? ( E1 or FXO card) sip.conf, zapata.cons, zaptel.conf config details?? Thanks Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +9466596 yahoo/skype Ids - vidurased == Message: 5 Date: Mon, 10 Dec 2007 15:26:52 -0800 From: Jai Rangi [EMAIL PROTECTED] Subject: [asterisk-users] Didnt get a frame from Channel and call gets disconnected To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hello, Since last few days I have noticed some people complaining that their call gets disconnected while they are in the middle of the conversations. Looking in the log I found these error messages, Dec 10 11:18:56 DEBUG[8833] channel.c: Bridge stops bridging channels SIP/5060-b7a03560 and SIP/219.206.2.291-089d8768 Dec 10 11:26:41 DEBUG[10410] channel.c: Didn't get a frame from channel: SIP/5060-b7a03560 Dec 10 11:26:41 DEBUG[10410] channel.c: Bridge stops bridging channels SIP/5060-b7a03560 and SIP/219.206.2.291-089d8768 Dec 10 11:26:53 DEBUG[10415] channel.c: Didn't get a frame from channel: SIP/5060-b7a0e2a8 Dec 10 11:26:53 DEBUG[10415] channel.c: Bridge stops bridging channels SIP/5060-b7a0e2a8 and SIP/Vendor-089d35b8 Dec 10 12:06:45 DEBUG[17210] channel.c: Didn't get a frame from channel: SIP/5060-b7a03560 Dec 10 12:06:45 DEBUG[17210] channel.c: Bridge stops bridging channels SIP/5060-b7a03560 and SIP/2219.206.2.291-089d8768 Dec 10 12:40:15 DEBUG[23089] channel.c: Didn't get a frame from channel: SIP/5060-b7a01728 Dec 10 12:40:15 DEBUG[23089] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Dec 10 12:57:48 DEBUG[25800] channel.c: Didn't get a frame from channel: SIP/5060-b7a01728 Dec 10 12:57:48 DEBUG[25800] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Dec 10 12:58:05 DEBUG[25809] channel.c: Didn't get a frame from channel: SIP/5060-b7a01728 Dec 10 12:58:05 DEBUG[25809] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Dec 10 14:10:36 DEBUG[5927] channel.c: Didn't get a frame from channel: SIP/5060-b7a01728 Dec 10 14:10:36 DEBUG[5927] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Dec 10 14:11:28 DEBUG[5961] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Dec 10 14:11:34 DEBUG[5961] channel.c: Didn't get a frame from channel: SIP/5060-b7a01728 Dec 10 14:11:34 DEBUG[5961] channel.c: Bridge stops bridging channels SIP/5060-b7a01728 and SIP/219.206.2.291-089d8768 Is this the right place to post this error message and expect for the solution. I am using asterisk-1.2.12 on FC5. I will appreciate if someone can give me some hints to get rid of this problem. Thank you, -JP ___ --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] Subject: Newb Question
Hi, Use orecx, voip call recording and monitoring. www.orecx.com Thanks Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +9466596 yahoo/skype Ids - vidurased -- Message: 17 Date: Fri, 30 Nov 2007 08:58:41 +0530 From: ram [EMAIL PROTECTED] Subject: Re: [asterisk-users] Newb Question To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 chan spy does the job i belive ram On Nov 30, 2007 7:37 AM, Jeff Adams [EMAIL PROTECTED] wrote: I inherited an office with phones that are hosted off-site. Everything is skinny and G729. I see that the FreeBSD asterisk port comes with a G729 codec. I want to record everything. If I use port mirroring on my switch, is it possible to configure asterisk to record and assemble packets that it doesn't otherwise route? Is it insane to user asterisk for this purpose? Advice or a link to a howto would be greatly 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
[asterisk-users] Re :Recommendations for 100 Wifi SIP phone
Hi, Try http://gigaset.siemens.com/shc/1,1935,hq_en_0_11729_rArNrNrNrN,00.html you will have good selections there. Regards, Vidura Senadeera To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 On Sun, 25 Nov 2007 23:26:54 +0300, [EMAIL PROTECTED] wrote: Hi all, Im preparing a quote for a 5 Star hotel, planning to have around 100 SIP Wifi phones for PBX operations running on 100 AccessPoints. Network is running in ARUBA Networks - AP70 access points. The initial recommendation is to go for Hitachi Wifiphones, but i would like to know from the group the recommendations. Im planning to put up Asterisk as the PBX, Please advice me the do's and donts as i'm not experienced on such heavy installation which are mission critical. I had been using asterisk on small profiles and this would be my first Pro setup with wifi handsets if all goes as planned. the Key Questions are Is Asterisk good enough? or do we need a another Proxy like SER? What is the experience with Hitachi Wifi phone's? Any specific Issues? Any such installations done? Please do a detail I had some Hitachi WIP5000 back in early 2006. It looks like a nice device but it really didn't deliver upon its promise. The reason to select a wifi phone is that by staying IP end to end you might gain operational advantages. It should have at least some of the features of a proper SIP deskset. The WIP5000 did not provide this at the time. The phones that I used had simple trouble moving between access points. Also the volume of the earpiece was very low, even for use in a quiet office. I am led to beleive that the new DECT cordless IP devices, like the system from Aastra Telecom, are currently a better option than wifi devices. Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --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 behind a PIX firewall?
Hi all, use ingate siparator. www.ingate.com ingate will help you to get rid of these issues. Regards, Vidura Senadeera Tel - +9466596 yahoo, skype - vidurased Sri Lanka. = You can also create the vpn using the existing pix and netgear, eliminating more hardware and points of failure. - Original Message - From: Ricardo Carvalho [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 27, 2007 7:30:35 AM (GMT-0800) America/Los_Angeles Subject: Re: [asterisk-users] Asterisk behind a PIX firewall? Try to just open port 5060 for SIP signaling on the PIX and also enable the INSPECT SIP rule. That way, your PIX firewall will inspect SIP signalling and open the necessary UDP ports for the RTP. If you have NAT uptream in the network, you should see if in the layer 4 the IPs shown in the SIP messages got rewritten by its public IPs, it should have, or else you'll never get it working right. Regards, Ricardo Carvalho. ___ --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] Vicidial + Unicall mfcr2
Hi Bruno, actually vicidial is working on top of asterisk, vicidial doesn't know what asterisk using in layer 2. SS7, ISDN stack, Unicall/mfcr2 is working with asterisk. vicidial uses asterisk application to deliver call center functionalities. Regards, Vidura. Dear Bruno, I had the experience of using the Vcidial with the boards of Digivoice. It worked very well! Leonardo Silva ___ --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] MFC/R2 protocol varient - sri lanka/Nortel DMS 100
Hi All, We successfully installed MFC/R2, chan_unicall.so with asterisk ver 1.2.6. asterisk is loading properly and we can see US show channels working fine. We are using digium Te120P card. Now we are trying to setup E1 link with Nortel DMS 100, which is resides at one of telco provider in Sri Lanka. But we don't know what is the exact protocol varient to use. Is anyone help us out on this reagard. How do we know the exact details of the protocol varient we have to use?. Thanks Regards, Vidura Senadeera, Senior solutions Specialist, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --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 Dropping Calls (Richard Young)
Hi,
Remove
usecallingpres=yes
busydetect=yes
from your zapata.conf file. and the restart asterisk. Hopefully you will not
faced drop call issues.
Regards,
Vidura Senadeera.
Message: 3
Date: Mon, 24 Sep 2007 12:29:40 +0100
From: Richard Young [EMAIL PROTECTED]
Subject: [asterisk-users] Asterisk Dropping Calls
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Hello,
I am having an issue whereby calls are being dropped randomly. I have an
ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk
install is based on Trixbox 2.0. However, I have updated the source code
to the following. The Asterisk release is asterisk-1.2.20. Zaptel
release is zaptel-1.2.18. And libpri release is libpri-1.2.4.
I have include an extract from the Asterisk log file below that shows
SIP/781 dropping a call when bridged to Zap/3-1. I have also included my
zaptel and zapata conf files.
I have researched the various messages displayed in the log file but
couldn't see anything that would point definitively to why calls are
being dropped.
Has anyone experienced anything similar or can anyone give me a few
ideas on where to start looking for the cause of the drop-outs?
Many thanks.
/var/log/asterisk/full:
Channel 0/3, span 1 got hangup request, cause 16
Sep 18 16:01:03 DEBUG[32377] channel.c: Didn't get a frame from channel:
Zap/3-1
Sep 18 16:01:03 DEBUG[32377] channel.c: Bridge stops bridging channels
SIP/781-b6e1b590 and Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Set option AUDIO MODE, value:
ON(1) on Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Hangup: channel: 3 index = 0,
normal = 15, callwait = -1, thirdcall = -1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Not yet hungup... Calling
hangup once with icause, and clearing call
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: disabled echo cancellation on
channel 3
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Updated conferencing on 3, with
0 conference users
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Set option AUDIO MODE, value:
OFF(0) on Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: disabled echo cancellation on
channel 3
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Hungup 'Zap/3-1'
Sep 18 16:01:03 DEBUG[32377] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Sep 18 16:01:03 VERBOSE[32377] logger.c: == Spawn extension
(macro-dialout-trunk, s, 20) exited non-zero on 'SIP/781-b6e1b590' in
macro 'dialout-trunk'
Sep 18 16:01:03 VERBOSE[32377] logger.c: == Spawn extension
(macro-dialout-trunk, s, 20) exited non-zero on 'SIP/781-b6e1b590'
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
Macro(SIP/781-b6e1b590, hangupcall) in new stack
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
ResetCDR(SIP/781-b6e1b590, w) in new stack
Sep 18 16:01:03 DEBUG[32377] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Sep 18 16:01:03 DEBUG[32377] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode) VALUES ('2007-09-18
15:58:30','02072900400','02072900400','08704440730','from-internal',
'SIP/781-b6e1b590','Zap/3-1','ResetCDR','w',153,150,'ANSWERED',3,'')
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: ResetCDR
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
NoCDR(SIP/781-b6e1b590, ) in new stack
Sep 18 16:01:03 NOTICE[32377] cdr.c: CDR on channel 'SIP/781-b6e1b590'
not posted
Sep 18 16:01:03 NOTICE[32377] cdr.c: CDR on channel 'SIP/781-b6e1b590'
lacks end
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: NoCDR
Sep 18 16:01:03 DEBUG[32377] pbx.c: Expression result is '1'
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
GotoIf(SIP/781-b6e1b590, 1?skiprg) in new stack
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Goto
(macro-hangupcall,s,6)
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: GotoIf
Sep 18 16:01:03 DEBUG[32377] pbx.c: Expression result is '1'
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
GotoIf(SIP/781-b6e1b590, 1?theend) in new stack
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Goto
(macro-hangupcall,s,9)
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: GotoIf
Sep 18 16:01:03 VERBOSE[32377] logger.c: -- Executing
Wait(SIP/781-b6e1b590, 5) in new stack
Sep 18 16:01:03 DEBUG[13856] chan_sip.c: Setting NAT on RTP to 524288
Sep 18 16:01:03 DEBUG[13856] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 1: Match
Found
Sep 18 16:01:03 DEBUG[13856] chan_sip.c: Setting NAT on RTP to 524288
My zaptel.conf is as follows:
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed
[asterisk-users] ztcfg error : TE110p error with CAS signalling on span 1 conflicts with HDLC with ...
Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. cas signalling on span1 conflicts with HDLC with FCS on channel 16 The switchtype and signalling im using is national, pri_cpe I'm attaching the avaya config details for more information. Please help me to sorted out this problem. - Thanks Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520036 Mobile - +9466596 SIGNALING GROUP Group Number: 1 Group Type: isdn-pri Associated Signaling? y Max number of NCA TSC: 0 Primary D-Channel: 01B0216 Max number of CA TSC: 0 Trunk Group for NCA TSC: Trunk Group for Channel Selection: X-Mobility/Wireless Type: NONE Supplementary Service Protocol: a DS1 CIRCUIT PACK Location: 01B02 Name: ZTE 1 Bit Rate: 2.048Line Coding: hdb3 Signaling Mode: isdn-pri Connect: network TN-C7 Long Timers? n Country Protocol: 7 Interworking Message: PROGress Interface Companding: alaw CRC? n Idle Code: DCP/Analog Bearer Capability: 3.1kHz Slip Detection? y Near-end CSU Type: other Echo Cancellation? n TRUNK GROUP Group Number: 1Group Type: isdn CDR Reports: y Group Name: OUTSIDE CALLCOR: 14 TN: 1TAC: 801 Direction: two-wayOutgoing Display? n Carrier Medium: PRI/BRI Dial Access? yBusy Threshold: 99Night Service: Queue Length: 0 Service Type: public-ntwrk Auth Code? nTestCall ITC: rest Far End Test Line No: TestCall BCC: 4 TRUNK PARAMETERS Codeset to Send Display: 6 Codeset to Send National IEs: 6 Max Message Size to Send: 260 Charge Advice: none Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc Trunk Hunt: cyclical Digital Loss Group: 13 Calling Number - Delete: Insert: Numbering Format: Bit Rate: 1200 Synchronization: sync Duplex: full Disconnect Supervision - In? y Out? n Answer Supervision Timeout: 0 TRUNK FEATURES ACA Assignment? nMeasured: none Wideband Support? n Maintenance Tests? y Data Restriction? n NCA-TSC Trunk Member: Send Name: n Send Calling Number: y Used for DCS? n Suppress # Outpulsing? nNumbering Format: public Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider Replace Restricted Numbers? y Replace Unavailable Numbers? y Send Connected Number: y Send UUI IE? y Send UCID? n Send Codeset 6/7 LAI IE? y Ds1 Echo Cancellation? n US NI Delayed Calling Name Update? n SBS? n Network (Japan) Needs Connect Before Disconnect? n DS1 CIRCUIT PACK Location: 01B01 Name: ZTE 4 Bit Rate: 2.048Line Coding: hdb3 Signaling Mode: isdn-pri Connect: network TN-C7 Long Timers? n Country Protocol: 7 Interworking Message: PROGress Interface Companding: alaw CRC? n Idle Code: DCP/Analog Bearer Capability: 3.1kHz Slip Detection? y Near-end CSU Type: other Echo Cancellation? n ___ --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] 14. Re: ztcfg error : TE110p error with CAS signalling on span1 conflicts with HDLC with ... (Carlos Chavez)
Hi Carlos/All, Thanks for your reply. I can remove dchan=16 from zaptel.conf But according to the documentation of Digium and sangoma they mentioning to use dchan=16. Are there any specific reason you have experiance regarding this and I am confusing that what this is included to the documentations. Regards, Vidura. On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote: Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 Remove dchan=16 from zaptel.conf. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.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] (no subject)
Motherboard with SATA RAID1 support That's a mulit-port SATA controller with RAID in the driver (software). 256 MB RAM Use a little more RAM. digium PRI/E1 card Is there any reason you aren't using Sangoma cards? 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) None. 2. If I use Hardware based RAID 1, what would be the impact to the system? A PCI slot. 3. According to your practical experiance what is the ideal solution among both options? Software RAID works fine. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk (Andrew Joakimsen)
Dear Andrew, Thanks for your kind responce. Regards, vidura. = Motherboard with SATA RAID1 support That's a mulit-port SATA controller with RAID in the driver (software). 256 MB RAM Use a little more RAM. digium PRI/E1 card Is there any reason you aren't using Sangoma cards? 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) None. 2. If I use Hardware based RAID 1, what would be the impact to the system? A PCI slot. 3. According to your practical experiance what is the ideal solution among both options? Software RAID works fine. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --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] TE210P digim card PRI problem
Hi, you have to correct your etc/zaptel.conf as follows span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 then try Regards, Vidura == I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-32 dchan=16 span=1,1,0,ccs,hdb3 bchan=32-46,49-62 dchan=47 in /etc/asterisk/zapata.conf switchtype=qsig context=zap-in signalling=pri_cpe group=1 channel = 1-15,17-31 group=2 channel = 32-46,49-62 is this configuration is fine or any other problem when i call to my second e1 which i connected to direct telcom i got this error call can't forward caz voice or dtmf can anyone send my runing configuration file of zaptel.conf or zapata.conf file waiting for your reply === ___ --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] Saftware RAID1 or Hardware RAID1 with Asterisk (Vidura Senadeera)
Dear all, Thanks for the greate explanation regaing Software/H/W Raid. This details better but on voip-info.org/wiki pages. Thanks lot agian. Regs, Vidura Senadeera. == Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? With my other hat on I build and maintain many servers with disk capacities ranging from 80GB to over 6TB... All using Linux software RAID. I've been using Linux s/w RAID for over 8 years now. So with RAID-1 done in hardware, the impact to the system, CPU, etc. should be no more (or less) than running a single SCSI or SATA drive. You write the data over the (PCI) bus once and the hardware takes care of writing it to both drives behind your back. Similarly for reading (where it might only read from one drive or from alternative drives) you only see one transaction over the PCI bus. You do (sometimes) need the hardware RAID controller to be supported by Linux and this is a weak area. Some controllers just look like a standard drive, so they are transparent to the system, but then you need to use either the BIOS utilities to set it up in the first place, or (typically) a Windows utility, although some controllers are now being supported by Linux with user-land tools to manage and check the arrays. Doing it in software requires double the PCI bandwidth for writes, but the same as a single drive or hardware controller for reads. AIUI, the current software RAID-1 reads alternatively from the disks. So on writes. The overhead in terms of CPU power is minimal - write the same block twice, and if the hardware is good, then both writes can be transfered over the PCI bus rapidly, into the cache on the drives and the writes then take place in parallel, so performance wise, it's really no worse than single drive (and it's important to note than it's no better than a single drive on reads too, despite many threads on the linux-raid list suggesting otherwise!) RAID-1 doesn't require parity calculations, so the software overhead really is quite small (especially when you compare it to the relatively huge times it takes to actually get the data to/from the disks) So things that are important: Make sure the hardware to each drive is as independent as possible. Hard to do these days as there is probably only one SATA controller chip on the motherboard. You also need to see what happens when a drive dies - is it going to crowbar the entire SATA chip and block the other drive? Is the driver going to recognise it quickly enough and so on. (Some early SATA drives weren't good at this) And the usual - make sure all the hardware has it's own interrupts. For the absolute maximun performance, (and minimum overheard) then you need a motherboard with multiple PCI buses - put the disks on one bus, the PRI card on another. If terms of disk b/w needed - if we're using g711, then it's 64KB/sec, and 20 calls streaming to voicemail is 1.3MB/sec. A single modern drive ought to be able to sustain 60MB/sec read or writes, so there is plenty of overhead, as long as asterisk is relatively sensible about buffering disk write/reads (which I think it is) So I'd say go for it, but do take the time, if possible to build a custom kernel for your hardware, and at the BIOS level, turn off all drivers that you won't be using - eg. on-board sound, then 2nd network port, USB (if you're not using it, don't enable it!) and so on, and make sure you have a custom compiled kernel for your exact hardware requirements with no modules loaded other than the Zap/TDM, etc., ones. And I'd also say go for it because I have similarly specd. servers doing similar tasks also running asterisk. I won't put a server in a remote data centre these days without it either booting off flash, or using at least RAID-1. Remember to put your swap on RAID-1 too. Here is one of my servers in a similar setup to yours: $ cat /proc/mdstat Personalities : [raid0] [raid1] md1 : active raid1 hdc1[1] hda1[0] 248896 blocks [2/2] [UU] md2 : active raid1 hdc2[1] hda2[0] 995904 blocks [2/2] [UU] md3 : active raid1 hdc3[1
[asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk
Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk server in production enviorment which is having E1 connection to the asterisk server, approximately 20 con-current calls, Music on hold, voice mail boxes. 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) 2. If I use Hardware based RAID 1, what would be the impact to the system? 3. According to your practical experiance what is the ideal solution among both options? I will be highly appreciate your feedback on this regard. -- Thanks Regards, Vidura Senadeera, ___ --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] Upgrade Asterisk
Hi, Try first installing latest release of libpri, then zaptel Try install asterisk after then. ope you will be able to compile it without any probs. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --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] Fwd: problem with one way audio
-- Forwarded message -- From: Vidura Senadeera [EMAIL PROTECTED] Date: Jun 27, 2007 1:56 PM Subject: Re: problem with one way audio To: asterisk-users@lists.digium.com Hi, If you have analog or digital cards installed. make sure to configure cards with proper signalling in /etc/zapel.conf. Hope you will be eliminate the issue using this hint. Regards, Vidura. ___ --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] TDM800P - zaptel service startup problem
Dear Team, I have installed digium TDM800P card. its include 1 quad fxo module and 2 FXO modules. I installed zaptel 1.2.18, libpri-1.2.4 and asterisk 1.2.19. I installed all zaptel drivers , asterisk without any problem. following are my /etc/zapel.conf settings fxsks=1,2,3,4,5,6 /etc/sysconfig/zaptel MODULES=$MODULES wctdm8xxp The proble is once i reboot the server, zaptel service failed. error message is : Loading zaptel framework: [ OK ] Waiting for zap to come online...Error: missing /dev/zap! Please give me a feedback on this regard. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recomender Server specs for 250 con-current calls
Dear All, I looking to implement asterisk solution for 2000 sip registrations and expecting con-current call about 250. Can some one provide me guide line that what kind of server will fullfil the requirment. what is the Processor, RAM ??? -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] busy/hangup/answer detection in PRI E1 channels
Hi, Please discribe me how we define busy/hang/answer detection with PRI E1 channels. Since busydetect, callprogress, busycount giving falts hangup and call drops what is the solution on PRI channels? -- Thanks Regards, Vidura B. Senadeera. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: busy/hangup/answer detection in PRI E1 channels (Vidura Senadeera)
Hi Gareth Blades Doug,
Thanks so much for for the feedback. I have searched on lot of documents
but couldn't able to find clear answer regarding it.
I hope you guys replies are very much help all in aterisk community.
Thanks Regards,
Vidura Senadeera,
Network Engineer,
Debug Solutions
Sri Lanka .
Tel - +94114520036
Mobile - +9466596
Web - www.debug.lk
--
Message: 14
Date: Thu, 15 Mar 2007 15:39:07 +0530
From: Vidura Senadeera [EMAIL PROTECTED]
Subject: [asterisk-users] busy/hangup/answer detection in PRI E1
channels
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Hi,
Please discribe me how we define busy/hang/answer detection with PRI E1
channels.
Since busydetect, callprogress, busycount giving falts hangup and call
drops
what is the solution on PRI channels?
--
Thanks Regards,
Vidura B. Senadeera.
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Message: 16
Date: Thu, 15 Mar 2007 10:35:16 +
From: Gareth Blades [EMAIL PROTECTED]
Subject: Re: [asterisk-users] busy/hangup/answer detection in PRI E1
channels
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain
You can use the hangupcause variable which us the pri cause code
supplied when a call is ended over a PRI line. For example this is the
maco we use to dial a number over PRI.
[macro-pridial]
exten = s,1,GotoIf($[${ARG1:0:2} != 00]?noint)
exten = s,n,Set(DENYINT=${DB(denyinternational/${CALLERIDNUM})})
exten = s,n,GotoIf($[ ${DENYINT} = yes ]?congestion)
exten = s,n(noint),Set(BLOCKCID=${DB(blockcid/${CALLERIDNUM})})
exten = s,n,GotoIf($[ ${BLOCKCID} = yes ]?prohib:cont)
exten = s,n(prohib),SetCallerPres(prohib)
exten = s,n(cont),Dial(ZAP/g1/${ARG1},60,Tr)
exten = s,n,Set(CDR(userfield)=${HANGUPCAUSE}.${DIALSTATUS})
exten = s,n,GotoIf($[ ${DIALSTATUS} = BUSY ]?busy)
exten = s,n,GotoIf($[ ${DIALSTATUS} = CONGESTION ]?congestion)
exten = s,n,GotoIf($[ ${HANGUPCAUSE} = 28 ]?unrecognised)
exten = s,n,GotoIf($[ ${HANGUPCAUSE} = 1 ]?discon)
exten = s,n,GotoIf($[ ${DIALSTATUS} = CHANUNAVAIL ]?congestion)
exten = s,n,Hangup
exten = s,n(busy),Busy
exten = s,n(congestion),GotoIf($[ ${HANGUPCAUSE} = 34 ]?error)
exten = s,n,Congestion
exten = s,n(error),Answer
exten = s,n,SendText(${HANGUPCAUSE}: ERROR: No channels available)
exten = s,n,Wait(1)
exten = s,n,Playback(all-outgoing-lines-unavailable)
exten = s,n,Wait(10)
exten = s,n,Hangup
exten = s,n(unrecognised),Answer
exten = s,n,SendText(${HANGUPCAUSE}: Unrecognised No.)
exten = s,n,Wait(1)
exten = s,n,Playback(that-is-not-rec-phn-num)
exten = s,n,Wait(10)
exten = s,n,Hangup
exten = s,n(discon),Answer
exten = s,n,SendText(${HANGUPCAUSE}:Out Of Service)
exten = s,n,Wait(1)
exten = s,n,Playback(discon-or-out-of-service)
exten = s,n,Wait(10)
exten = s,n,Hangup
On Thu, 2007-03-15 at 10:09, Vidura Senadeera wrote:
Hi,
Please discribe me how we define busy/hang/answer detection with PRI
E1 channels.
Since busydetect, callprogress, busycount giving falts hangup and call
drops what is the solution on PRI channels?
--
Thanks Regards,
Vidura B. Senadeera.
Message: 18
Date: Thu, 15 Mar 2007 07:06:30 -0400
From: Doug Lytle [EMAIL PROTECTED]
Subject: Re: [asterisk-users] busy/hangup/answer detection in PRI E1
channels
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
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Vidura Senadeera wrote:
Hi,
Please discribe me how we define busy/hang/answer detection with PRI
E1 channels.
Since busydetect, callprogress, busycount giving falts hangup and call
drops what is the solution on PRI channels?
PRI channels have call supervision and Asterisk will see the
hangup/answers just fine. The busydetect, callprogress, busycount
should be removed from your setup.
Doug
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[asterisk-users] Re: Back to back E1 - asterisk = toshiba pbx - Call droping issue [ ref:00D36mPe.50032wycQ:ref ]
Hi All, Thanks for every one who helped me on this regard. I think i was able to rictify the problem. what i did is remove callprogress=yes usecallinpres=yes and restart asterisk. Today i didn't report any drop calls. Many thanks for Eric. :) I hope this situation will continue. Regards, Vidura. On 3/8/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Hi, Opps ...there are some more attachments i missed to send you. Please refer. sorry for the inconvenience occured. Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk On 3/8/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Hi, Today I have reported 10 calls drop within 2.5 hours period of time. This is being a huge issue. I'm using Asterisk 1.2.14 and zaptel 1.2.12 and libpri-1.2.4. Pls find attached files you have requested. Thanks, Vidura. On 3/8/07, Digium Support [EMAIL PROTECTED] wrote: Hi, Please make sure you are running the latest 1.2 or 1.4 stable releases of Asterisk/Zaptel and Libpri. Also could you send me the output of these two commands: cat /proc/interrupts lspci -bv Please let us know if you have any questions. -Regards David Faulk Digium Support Technician Digium Certified Asterisk Professional Digium, Inc. 150 West Park Loop, Suite 100 Huntsville, AL 35806 +1-256-428-6000 www.digium.com ref:00D36mPe.50032wycQ:ref -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pritimer parameter in zapata.conf
Hi all, Please discribe me more about pritimer parameter in zapata.conf http://lists.digium.com/pipermail/asterisk-commits/2006-July/005824.html I found above url and have some idea. My PRI E1 timer is t203, what is the best vale that i have to use for as counter. default is 1ms, If i changed it to some big amount, like 6 what will happen T203: Layer 2 max time without frames being exchanged (default 1 ms) -- Thanks Regards, Vidura B. Senadeera. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Back to back E1 - asterisk = toshiba pbx - Call droping
Hi steve and All, I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf, zaptel.conf for your information Thanks so much for the feedback and I do accordingly. Hope to get rid off this isue any how. To day also reported 10 call drops within 2 hours of period. fook forward to have your support on this regard. Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk Message: 16 Date: Wed, 7 Mar 2007 05:05:36 -0500 From: Steve Totaro [EMAIL PROTECTED] Subject: RE: [asterisk-users] Back to back E1 - asterisk = toshiba pbx - Calldroping issue To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii As these problems are very time sensitive and frustrating, I suggest you document each change you make and do them one at a time so you can actually know what the problem was and not introduce new problems in the process. Find someone who is on the phone quite a bit and will give you an honest evaluation of the call dropping situation (unless you yourself are experiencing this issue too). Some people are so quick to say, It is still happening without starting the evaluation from a clean slate after each change. You may want to check your Asterisk log for more insight. /var/log/asterisk/full. Also you can turn on debugging on one span at a time and see if you can find something there Do you have a resetinterval set in zapata.conf? If you can isolate the dropped calls to the reset interval (watch the console, it will scroll with each channel being reset) then set resetinterval=never. If there is no entry for resetinterval, add it and set it to never since it is defaulted to on. Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. This in combination with your first span should accept timing from the Telco and then supply it to your Toshiba, I would actually try this first. Another thought, It seems you have quite a lot of hardware in that box. I am not sure how much is too much, but that would probably just rear it's ugly head as poor audio. Thanks, Steve Totaro http://www.asteriskhelpdesk.com _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vidura Senadeera Sent: Wednesday, March 07, 2007 2:15 AM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: [asterisk-users] Back to back E1 - asterisk = toshiba pbx - Calldroping issue Hi Team, I have integrated asterisk with Toshiba analog PBX. NOw the live setup is going. Now I am facing call droping problem. It's happening ample time. 10-20 calls are droping every day. What could be the reason. I attached latest zapata.conf file for your information. This is being a huge issue. Highly appreciate your help on this regard. Thanks Regards, Vidura Senadeera. On 1/26/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear Marco, There is a huge problem i'm facing. My asterisk server included with TDM2451E and 2 TE110p cards. One E1 i conected to the telco. other E1 port i'm using to cros-connection with toshiba pbx. My telco E1 d channels communicating well. but toshiba pbx E1 not getting. d-channels are not getting up. what could be the issue. i'm using asterisk -1.2.14 and zaptel 1.2.12. notes - if i put, zap show channels in asterisk cli. its only showing the first 31 channels. but with ztcfg -vvv it showing al the channels. my configs are # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS/CRC4 RED # Suntel E1 connection == span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 # Span 2: WCT1/1 Digium Wildcard TE110P T1/E1 Card 1 # Legacy PBX E1 connection === span=2,2,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 # Span 3: WCTDM/0 Wildcard TDM2400P Prototype Board 1 fxoks=63 fxoks=64 fxoks=65 fxoks=66 fxoks=67 fxoks=68 fxoks=69 fxoks=70 fxoks=71 fxoks=72 fxoks=73 fxoks=74 fxoks=75 fxoks=76 fxoks=77 fxoks=78 fxoks=79 fxoks=80 fxoks=81 fxoks=82 fxsks=83 fxsks=84 fxsks=85 fxsks=86 # Global data loadzone= us defaultzone = us Regards, vidura -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070307/62a82c69/attachment-0001.htm -- Message: 17 Date: Wed, 7 Mar 2007 11:17:07 +0100 From: Thomas Deillon [EMAIL PROTECTED] Subject: [asterisk-users] Asterisk 1.4.1 - Calling problem To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi all, I
[asterisk-users] Fwd: Back to back E1 - asterisk = toshiba pbx - Call droping
-- Forwarded message -- From: Vidura Senadeera [EMAIL PROTECTED] Date: Mar 8, 2007 11:27 AM Subject: Re: Back to back E1 - asterisk = toshiba pbx - Call droping To: asterisk-users@lists.digium.com Hi steve and All, I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf, zaptel.conf for your information Thanks so much for the feedback and I do accordingly. Hope to get rid off this isue any how. To day also reported 10 call drops within 2 hours of period. fook forward to have your support on this regard. Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk Message: 16 Date: Wed, 7 Mar 2007 05:05:36 -0500 From: Steve Totaro [EMAIL PROTECTED] Subject: RE: [asterisk-users] Back to back E1 - asterisk = toshiba pbx - Calldroping issue To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii As these problems are very time sensitive and frustrating, I suggest you document each change you make and do them one at a time so you can actually know what the problem was and not introduce new problems in the process. Find someone who is on the phone quite a bit and will give you an honest evaluation of the call dropping situation (unless you yourself are experiencing this issue too). Some people are so quick to say, It is still happening without starting the evaluation from a clean slate after each change. You may want to check your Asterisk log for more insight. /var/log/asterisk/full. Also you can turn on debugging on one span at a time and see if you can find something there Do you have a resetinterval set in zapata.conf? If you can isolate the dropped calls to the reset interval (watch the console, it will scroll with each channel being reset) then set resetinterval=never. If there is no entry for resetinterval, add it and set it to never since it is defaulted to on. Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. This in combination with your first span should accept timing from the Telco and then supply it to your Toshiba, I would actually try this first. Another thought, It seems you have quite a lot of hardware in that box. I am not sure how much is too much, but that would probably just rear it's ugly head as poor audio. Thanks, Steve Totaro http://www.asteriskhelpdesk.com _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vidura Senadeera Sent: Wednesday, March 07, 2007 2:15 AM To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: [asterisk-users] Back to back E1 - asterisk = toshiba pbx - Calldroping issue Hi Team, I have integrated asterisk with Toshiba analog PBX. NOw the live setup is going. Now I am facing call droping problem. It's happening ample time. 10-20 calls are droping every day. What could be the reason. I attached latest zapata.conf file for your information. This is being a huge issue. Highly appreciate your help on this regard. Thanks Regards, Vidura Senadeera. On 1/26/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear Marco, There is a huge problem i'm facing. My asterisk server included with TDM2451E and 2 TE110p cards. One E1 i conected to the telco. other E1 port i'm using to cros-connection with toshiba pbx. My telco E1 d channels communicating well. but toshiba pbx E1 not getting. d-channels are not getting up. what could be the issue. i'm using asterisk -1.2.14 and zaptel 1.2.12. notes - if i put, zap show channels in asterisk cli. its only showing the first 31 channels. but with ztcfg -vvv it showing al the channels. my configs are # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS/CRC4 RED # Suntel E1 connection == span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 # Span 2: WCT1/1 Digium Wildcard TE110P T1/E1 Card 1 # Legacy PBX E1 connection === span=2,2,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 # Span 3: WCTDM/0 Wildcard TDM2400P Prototype Board 1 fxoks=63 fxoks=64 fxoks=65 fxoks=66 fxoks=67 fxoks=68 fxoks=69 fxoks=70 fxoks=71 fxoks=72 fxoks=73 fxoks=74 fxoks=75 fxoks=76 fxoks=77 fxoks=78 fxoks=79 fxoks=80 fxoks=81 fxoks=82 fxsks=83 fxsks=84 fxsks=85 fxsks=86 # Global data loadzone= us defaultzone = us Regards, vidura -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070307/62a82c69/attachment-0001.htm -- Message: 17 Date: Wed, 7 Mar 2007 11:17:07 +0100 From: Thomas Deillon [EMAIL PROTECTED] Subject: [asterisk-users] Asterisk 1.4.1
[asterisk-users] Back to back E1 - asterisk = toshiba pbx - Call droping issue
Hi Team, I have integrated asterisk with Toshiba analog PBX. NOw the live setup is going. Now I am facing call droping problem. It's happening ample time. 10-20 calls are droping every day. What could be the reason. I attached latest zapata.conf file for your information. This is being a huge issue. Highly appreciate your help on this regard. Thanks Regards, Vidura Senadeera. On 1/26/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Dear Marco, There is a huge problem i'm facing. My asterisk server included with TDM2451E and 2 TE110p cards. One E1 i conected to the telco. other E1 port i'm using to cros-connection with toshiba pbx. My telco E1 d channels communicating well. but toshiba pbx E1 not getting. d-channels are not getting up. what could be the issue. i'm using asterisk -1.2.14 and zaptel 1.2.12. notes - if i put, zap show channels in asterisk cli. its only showing the first 31 channels. but with ztcfg -vvv it showing al the channels. my configs are # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS/CRC4 RED # Suntel E1 connection == span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 # Span 2: WCT1/1 Digium Wildcard TE110P T1/E1 Card 1 # Legacy PBX E1 connection === span=2,2,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 # Span 3: WCTDM/0 Wildcard TDM2400P Prototype Board 1 fxoks=63 fxoks=64 fxoks=65 fxoks=66 fxoks=67 fxoks=68 fxoks=69 fxoks=70 fxoks=71 fxoks=72 fxoks=73 fxoks=74 fxoks=75 fxoks=76 fxoks=77 fxoks=78 fxoks=79 fxoks=80 fxoks=81 fxoks=82 fxsks=83 fxsks=84 fxsks=85 fxsks=86 # Global data loadzone= us defaultzone = us Regards, vidura -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. zapata.conf Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Integrating asterisk with Toshiba Astrata DK380
Deat all,
I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.
Following is my setup
*Asterisk - Digium TE110P - E1 card in toshiba pbx - Toshiba PBX*
A = B
C D
Asterisk PBX and strata PBX connected using back to back E1 cross cable.
Physicall connectivity is OK. The digium TE110p
LED state green. zttool also OK.
Toshiba stata configured to make outbound call via E1 link with pressing 9
and then the out side number.
I was able to make call from soft phone to analog extension at toshiba pbx.
A== B way as shown above. But when trying to dial from
Toshiba PBX analog extension to asterisk extension, by pressing 9 the call
rejected.
In the asterisk command prompt I'm having following error message.
-- Extension '' in context 'from-pstn' from '' does not exist. Rejecting
call on channel 0/31, span 1
Is there any wrong in my setup. dial plan??, additional configuration if i
required to put please guide me.
I will be greately appreciated your feedback on this regard.
*configuration details*
*/etc/zaptel.conf*
# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
*/etc/asterisk/zapata.conf*
signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=euroisdn
;switchtype=national
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number is
in milliseconds
callerid=asreceived
overlapdial=no
pridialplan=unknown
immediate=no
;rxwink=300
callprogress=no
loadzone=au
context=from-pstn ; Points to the default context of your extensions.conf
group=2
channel=1-15
channel=17-31 ;PRI/E1 link
[trunkgroups]
trunkgroup=2,16
spanmap=1,2,1
*/etc/asterisk/extension.conf*
[from-zaptel]
exten = _X.,1,Set(DID=${EXTEN})
exten = _X.,n,Goto(s,1)
exten = s,1,NoOp(Entering from-zaptel with DID == ${DID})
; If ($did == ) { $did = s; }
exten = s,n,Set(DID=${IF($[${DID}= ]?s:${DID})})
exten = s,n,NoOp(DID is now ${DID})
exten = s,n,GotoIf($[${CHANNEL:0:3}=Zap]?zapok:notzap)
exten = s,n(notzap),Goto(ext-did,${DID},1)
; If there's no ext-did,s,1, that means there's not a no did/no cid route.
Hangup.
exten = s,n,Macro(hangup)
exten = s,n(zapok),NoOp(Is a Zaptel Channel)
exten = s,n,Set(CHAN=${CHANNEL:4})
exten = s,n,Set(CHAN=${CUT(CHAN,-,1)})
exten = s,n,Macro(from-zaptel-${CHAN},${DID},1)
; If nothing there, then treat it as a DID
exten = s,n,NoOp(Returned from Macro from-zaptel-${CHAN})
exten = s,n,Goto(ext-did,${DID},1)
--
Thanks Regards,
Vidura B. Senadeera.
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[asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 79
Hi,
I checked by changing to from-zaptel, but no luck yet. Pls guide me on
this.
Regards,
vudura senadeera
--
Message: 9
Date: Fri, 19 Jan 2007 16:47:18 -
From: Robert Jenkins [EMAIL PROTECTED]
Subject: RE: [asterisk-users] Integrating asterisk with Toshiba
Astrata DK380
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Hi,
your zapata.con has 'context=from-pstn'
Try changing this to 'context=from-zaptel'
Robert Jenkins.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: 19 January 2007 15:19
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Subject: [asterisk-users] Integrating asterisk with Toshiba Astrata
DK380
Deat all,
I am in middle of integrate Asterisk with Toshiba astrata legacy pbx.
Following is my setup
Asterisk - Digium TE110P - E1 card in toshiba pbx - Toshiba PBX
A = B
C D
Asterisk PBX and strata PBX connected using back to back E1 cross cable.
Physicall connectivity is OK. The digium TE110p
LED state green. zttool also OK.
Toshiba stata configured to make outbound call via E1 link with pressing
9
and then the out side number.
I was able to make call from soft phone to analog extension at toshiba
pbx.
A== B way as shown above. But when trying to dial from
Toshiba PBX analog extension to asterisk extension, by pressing 9 the
call
rejected.
In the asterisk command prompt I'm having following error message.
-- Extension '' in context 'from-pstn' from '' does not
exist. Rejecting
call on channel 0/31, span 1
Is there any wrong in my setup. dial plan??, additional configuration if
i
required to put please guide me.
I will be greately appreciated your feedback on this regard.
configuration details
/etc/zaptel.conf
# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
/etc/asterisk/zapata.conf
signalling=pri_net ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=euroisdn
;switchtype=national
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your
needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number
is
in milliseconds
callerid=asreceived
overlapdial=no
pridialplan=unknown
immediate=no
;rxwink=300
callprogress=no
loadzone=au
context=from-pstn ; Points to the default context of your
extensions.conf
group=2
channel=1-15
channel=17-31 ;PRI/E1 link
[trunkgroups]
trunkgroup=2,16
spanmap=1,2,1
/etc/asterisk/extension.conf
[from-zaptel]
exten = _X.,1,Set(DID=${EXTEN})
exten = _X.,n,Goto(s,1)
exten = s,1,NoOp(Entering from-zaptel with DID == ${DID})
; If ($did == ) { $did = s; }
exten = s,n,Set(DID=${IF($[${DID}= ]?s:${DID})})
exten = s,n,NoOp(DID is now ${DID})
exten = s,n,GotoIf($[${CHANNEL:0:3}=Zap]?zapok:notzap)
exten = s,n(notzap),Goto(ext-did,${DID},1)
; If there's no ext-did,s,1, that means there's not a no did/no cid
route.
Hangup.
exten = s,n,Macro(hangup)
exten = s,n(zapok),NoOp(Is a Zaptel Channel)
exten = s,n,Set(CHAN=${CHANNEL:4})
exten = s,n,Set(CHAN=${CUT(CHAN,-,1)})
exten = s,n,Macro(from-zaptel-${CHAN},${DID},1)
; If nothing there, then treat it as a DID
exten = s,n,NoOp(Returned from Macro from-zaptel-${CHAN})
exten = s,n,Goto(ext-did,${DID},1)
--
Thanks Regards,
Vidura B. Senadeera.
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Message: 10
Date: Fri, 19 Jan 2007 11:46:57 -0500
From: Chris Earle \(CBL\) [EMAIL PROTECTED]
Subject: [asterisk-users] Disconnect Supervision UK / BT solution?
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Hi all
I'm using sangoma a200 cards in the UK and have the ongoing, often noted
problem of disconnect supervision with BT POTS lines.
Just noticed this post on
http://www.voip-info.org/wiki/view/UK+Asterisk+Details
stating that potentially someone's got a solution :
TDM400P amp; Not Detecting Hangups:
Got a TDM400P installed and having problems with Asterisk not detecting
hangups? Using BT? If so, contact BT and ask what the Disconnect Clear
Time setting is for your phone line. Odds are it's probably 100.
Increasing
it to 800 fixed the issue for me.
Disconnect Clear Time is BT's name for CPC.
Does anyone have any thoughts/confirmation about this finally being a
viable
solution? This disconnect supervision problem has plagued TDM and
Sangoma
cards for a long time!
Comments appreciated before I get on the phone with BT
[asterisk-users] Asterisk Developers Mailing List asterisk-dev@lists.digium.com,
Hi, I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are 1.2.11 and 1.2.3. Please refer the attached txt files for Linux version information and output of zaptel compile. I will be highly appreciated that any one canhelp meon this regard.-- Thanks Regards,Vidura B. Senadeera. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel-1.2.8 compile problem
Hi, I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are 1.2.11 and 1.2.3. Please refer the attached txt files for Linux version information and output of zaptel compile. I will be highly appreciated that any one canhelp meon this regard.-- Thanks Regards,Vidura B. Senadeera. -- Thanks Regards,Vidura B. Senadeera. cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits tones.h cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREmakefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h ./makefw pciradio.rbt radfw radfw.h ZAPTELVERSION=1.2.8 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp cc fw2h.c -o fw2h ./fw2h OCT6114-128D.ima vpm450m_fw.h cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWAREzttest.c -o zttest cc -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DHOTPLUG_FIRMWARE -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.EL/build make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/home/vidura/zaptel-1.2.8 modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /home/vidura/zaptel-1.2.8/zaptel.o make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-i686' Linux version 2.6.9-34.EL ([EMAIL PROTECTED]) (gcc version 3.4.5 20051201 (Red Hat 3.4.5-2)) #1 Wed Mar 8 00:07:35 CST 2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Integrate asterisk with Database
Hi All, I am plainging to give a solutions for a sports club. Follwing is the process that i need to achieve. If any body achieve this kind of setup pls give me a feedback, so that i can go through. Call flow start [for database operations please use an access database with suitably configured fields] Thank you for calling Sports World Press 1 for English, 2 for French Please key in your 12 digit membership number Your membership number is [repeat the digits that have been keyed in] Press 1 to confirm or 2 to key again [loop until confirmed] [exit after three invalid attempts and say] Invalid membership number. Please call customer services. [Access database and lookup the membership number and check validity field in database] [If missing or invalid bin] The membership number is invalid. Transaction terminated. [exit at this point] Please enter your mobile number Your mobile number is [repeat the digits that have been keyed in] Press 1 to confirm or 2 to key again [loop until confirmed] Please enter your activity code [4 digits] Your activity code is [repeat the digits that have been keyed in] Press 1 to confirm or 2 to key again [loop until confirmed] Please enter your activity duration [upto 5 digits] Your activity duration is [repeat the digits that have been keyed in] Press 1 to confirm or 2 to key again [loop until confirmed] [store mobile number, activity code and activity duration in the database] Your transaction is complete. Thank you for using sports world. ..Call flow end .. Thanks Regards, Vidura Senadeera. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intergrate Asterisk IP PBX with Legacy PBX, continuing existing funtionality of legacy pbx
Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story This is with regard to the setup which you can find at the Asterisk The Future of Telephony , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our Conventional PABX system. *Existing Setup over view* Our existing includes traditional Pabx, E1 Line from telecom provider, 16 direct lines another telecom provider. there are around 120 extensions. E1 Link using for DID and 16 lines using for as hunting group. *New Integration.* Integrate asterisk ip PBX with legacy Pabx which continues functionality of the existing setup I am planning to install 2 E1 cards in Asterisk box. Remove E1 link from legacy Pabx and fix it to 1 E1 card and other E1 card will using to connect traditional PABX. All previous DID's which configured with the traditional PABX will be configured in asterisk. Actually I am not sure that i will be able to achive this migration , but i am trying to acomplish. It is very much appreciate that if anyone can guide me on this regard. Thanks Regards, Vidura Senadeera. Sri Lanka. attachment: legacy_pbx_to_asterisk_migration.JPG ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
