Re: [asterisk-users] the lenght of the uri affects on dialplan?
mention the complete scnario and your sip.conf. Regards, Faisal (sent from phone) Rafael Visser rafael_vis...@hotmail.com wrote: Hi Gurus.. I use asterisk for just for ivr. My issue is that when the switch changes it's host name from MSSASU1.MYDOMAIN to MSSSASU1.MYDOMAiN.COM or MSSSASU1.MYDOMAiN.COM.PY the call is rejected with No matching peer and the handle_request_invite: Sending fake auth rejection for device x. It doesn't match it's own default context. Also, it has somethig to do with the numbers of digits of the dialed number. Few digits works ok, 14 to more works wrong. Do you know what am i missing? Thanks in advance. Debug with long hostname (B is considered as an '*') --- SIP read from TCP:10.146.9.70:6240 --- INVITE sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone SIP/2.0 From: sip:971200...@mssasu1.mydomain.com.py;user=phone;tag=3016589695 To: sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone Max-Forwards: 70 Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK0035391821780096 Call-ID: 9cax8060616182201-bo...@mssasu1.mydomain.com.py CSeq: 7313 INVITE P-Asserted-Identity: sip:971200...@mssasu1.mydomain.com.py;user=phone Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE P-Charging-Vector: icid-value=6510081000-0826-16155907;icid-generated-at=MSSASU1.MYDOMAIN.COM.PY;orig-ioi=MSSASU1.MYDOMAIN.COM.PY Supported: 100rel Content-Type: application/sdp Contact: sip:MSSASU1.MYDOMAIN.COM.PY:5060;transport=UDP Content-Length: 414 v=0 o=- 7530078 7530078 IN IP4 MSSASU1.MYDOMAIN.COM.PY s=- t=0 0 a=sendrecv m=audio 13802 RTP/AVP 8 96 18 97 c=IN IP4 10.143.1.67 b=RR:0 b=RS:0 a=rtpmap:8 PCMA/8000 a=rtpmap:96 AMR/8000 a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=maxptime:40 - --- (15 headers 17 lines) --- Sending to 10.146.9.70:5060 (no NAT) Using INVITE request as basis request - 9cax8060616182201-bo...@mssasu1.mydomain.com.py No matching peer for '971200152' from '10.146.9.70:6240' [Aug 26 16:15:55] NOTICE[6873]: chan_sip.c:21975 handle_request_invite: Sending fake auth rej ection for device sip:971200...@mssasu1.mydomain.com.py;user=phone;tag=3016589695 # --- Reliably Transmitting (no NAT) to 10.146.9.70:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP MSSASU1.MYDOMAIN.COM.PY:5060;branch=z9hG4bK0035391821780096;received=10.146.9.70 From: sip:971200...@mssasu1.mydomain.com.py;user=phone;tag=3016589695 To: sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone;tag=as4cfd0d54 Call-ID: 9cax8060616182201-bo...@mssasu1.mydomain.com.py CSeq: 7313 INVITE Server: Asterisk PBX 1.8.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=35ff0feb Content-Length: 0 Short hostname on switch === Connected to Asterisk 1.8.7.0 currently running on fdosis-ims1 (pid = 25430) fdosis-ims1*CLI core set verbose 1 Verbosity was 0 and is now 1 --- SIP read from UDP:10.146.9.70:5060 --- INVITE sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone SIP/2.0 From: sip:971200152@MSSASU1.MYDOMAIN;user=phone;tag=0046120455 To: sip:b56510123456789012...@sisivr03.mydomain.com.py;user=phone Max-Forwards: 70 Via: SIP/2.0/UDP MSSASU1.MYDOMAIN:5060;branch=z9hG4bK0038670956791982 Call-ID: qDaQ1240646182201-AKDE-@MSSASU1.MYDOMAIN CSeq: 14481 INVITE P-Asserted-Identity: sip:971200152@MSSASU1.MYDOMAIN;user=phone Accept: application/sdp llow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE P-Charging-Vector: icid-value=A6D6B81000-0826-16440101;icid-generated-at=MSSASU1.MYDOMAIN;orig-ioi=MSSASU1.MYDOMAIN.COM.PY Supported: 100rel Content-Type: application/sdp Contact: sip:MSSASU1.MYDOMAIN:5060;transport=UDP Content-Length: 407 v=0 o=- 8986991 8986991 IN IP4 MSSASU1.MYDOMAIN s=- t=0 0 a=sendrecv m=audio 30838 RTP/AVP 8 96 18 97 c=IN IP4 10.143.1.68 b=RR:0 b=RS:0 a=rtpmap:8 PCMA/8000 a=rtpmap:96 AMR/8000 a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=maxptime:40 - --- (15 headers 17 lines) --- Sending to 10.146.9.70:5060 (no NAT) Using INVITE request as basis request - qDaQ1240646182201-AKDE-@MSSASU1.MYDOMAIN Found peer 'sip.ericsson' for '971200152' from 10.146.9.70:5060 Found RTP audio format 8 Found RTP audio format 96 Found RTP audio format 18 Found RTP audio format 97 Found audio description format PCMA for ID 8 Found unknown media description format AMR for ID 96 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 97 Capabilities: us - 0xe
Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?
hi, you can simply avoid this by using local ring r option in dial command. azterisk pass local ring voice to caller and will not bridge b leg audio until b leg is answered.iin Regards, Faisal Hanif (sent from phone) Steve Davies davies...@gmail.com wrote: Hi SIP Gurus, I've tried to find the relevant RFCs, but am struggling. I can find the odd opinion online, but was wondering if anyone could give a definitive answer. If a SIP call is initiated (INVITE) and receives either a 180 with SDP, or a 183 with SDP, then the remote party will start to send audio for the inband-ringing. Asterisk then passes this audio, and it is correctly heard by the caller. At present, Asterisk will also start to pass back any handset audio in return, in theory allowing a conversation to occur on an unanswered channel if an endpoint were designed to allow this (free phonecalls here we come!). My question: Should: 1) Asterisk block outbound audio between the 183 Progress and the 200 OK packets? 2) Replace any outbound audio with silence? 3) Replace outbound audio with a special NULL RTP of some sort? Does that exist? 4) Allow any audio to be sent regardless? I have implemented 1) at present on our test rig, but the lack of outbound RTP causes issues with firewall state not being set-up to allow the inbound audio. I am not sure how hard/easy it would be to do 2) as you'd need to create silence of the correct duration to replace each audio frame. Thoughts please? Many thanks, Steve -- _ -- 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 -- _ -- 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
Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?
You can create trunk/route specific dial command parameters. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies Sent: Friday, August 24, 2012 8:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Question - Early audio one-way or 2-way? On 24 August 2012 15:34, Faisal Hanif fai...@vopium.com wrote: Steve Davies davies...@gmail.com wrote: Hi SIP Gurus, I've tried to find the relevant RFCs, but am struggling. I can find the odd opinion online, but was wondering if anyone could give a definitive answer. If a SIP call is initiated (INVITE) and receives either a 180 with SDP, or a 183 with SDP, then the remote party will start to send audio for the inband-ringing. Asterisk then passes this audio, and it is correctly heard by the caller. At present, Asterisk will also start to pass back any handset audio in return, in theory allowing a conversation to occur on an unanswered channel if an endpoint were designed to allow this (free phonecalls here we come!). My question: Should: 1) Asterisk block outbound audio between the 183 Progress and the 200 OK packets? 2) Replace any outbound audio with silence? 3) Replace outbound audio with a special NULL RTP of some sort? Does that exist? 4) Allow any audio to be sent regardless? I have implemented 1) at present on our test rig, but the lack of outbound RTP causes issues with firewall state not being set-up to allow the inbound audio. I am not sure how hard/easy it would be to do 2) as you'd need to create silence of the correct duration to replace each audio frame. Thoughts please? Many thanks, Steve hi, you can simply avoid this by using local ring r option in dial command. azterisk pass local ring voice to caller and will not bridge b leg audio until b leg is answered.iin Regards, Faisal Hanif (sent from phone) Nice thought, but what if there is a useful reason for the progress audio? If it is sent we want to hono[u]r it, and if it is not, we expect a 180 ringing, and let the SIP device generate the tone, rather than send an unwanted audio stream to use up bandwidth :) For example, some UK ISDN services will give a useful call failure message by sending a progress-frame, followed by some audio. DAHDI and SIP handle this nicely with a 183 progress message, and pass on the message on the un-answered SIP channel. Regards, Steve -- _ -- 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 -- _ -- 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
Re: [asterisk-users] CDRs on multiple servers.
The easiest way for you to use MySQL-Relay or MySQL-Proxy with ODBC. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Owais Ahmad Sent: Tuesday, June 05, 2012 7:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDRs on multiple servers. Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. Thanks -- _ -- 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 -- _ -- 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
Re: [asterisk-users] sip show peers
If I understand correct you need to increase qualify value. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Tuesday, May 22, 2012 5:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] sip show peers I have a process that runs on a server and does a simple 'asterisk -rx sup show peers' /tmp/peers and then looks for any (Unspecified) items and reports them as having lost connection. My server is running 1.4.43 and the two boxes I am monitoring are also running 1.4.43. Once in a great while 1 of my boxes reports (Unspecified). I am trying to find out why. How can I make the remote boxes have a shorter heart beat to checking more frequently with the server so as not to go (Unspecified). By the time I log in and check its already back connected again. Any other thoughts? Thanks, Jerry -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Multiple route failover zaps registration
Why don't you use FQDN in phone instead of IP of server and configure DNS Server to failover resolve to next IP while set SIP reg expiry same as DNS TTL. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Monday, December 12, 2011 5:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Multiple route failover zaps registration Hi all, I've got a customer who is bringing up a second Internet connection for fail- over. I've configured a WRT54 with 2 LAN ports and arranged for it to fail over when one of the routes is no longer available. That works just fine at the IP level. However, when the router fails over, the phones lose their registration, presumably because their IP address has changed from Asterisk's point of view. The phones happen to be Polycom 335's, and I'm running Asterisk 1.6.2.9. What is the best way to manage this situation so that the phones don't become unavailable during failover? I'm considering using the Tinc VPN solution to prevent the IP address from chaing, but I'm hoping for a more simple solution. Any ideas? -- Take care and have fun, Mike Diehl. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Best VoIP conferencing phone ?
I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. Regards, Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 30, 2011 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind Subject: [asterisk-users] Best VoIP conferencing phone ? Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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
Re: [asterisk-users] Best VoIP conferencing phone ?
In hardware I used some snom phones up to six lines. You can check on http://www.snom.com/ for appropriate model. Regards, Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 30, 2011 1:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best VoIP conferencing phone ? Hi Faisal, Thanks for reply but I want hardware wase VoIP device. If know please gussed me. From google I fould the list of below devices but I am not sure that these are best for used or have an issue 1)Polycom SoundStation IP 7000 Why it's best: The Polycom SoundStation IP 7000 is the most advanced conference phone from the Polycom SoundStation lineup and leaves little to be desired. With an amazing 20' 360 radius, the 7000 is perfect for large conference rooms. The new HD voice quality (22 kHz) allows. 2) Polycom Voicestation 500 Why it's a best pick: The Polycom VoiceStation 500 is one of the best conference phones for a wide variety of reasons. The VoiceStation 500 features amazing call quality, 7' 360 radius, Bluetooth connectivity, wired connection, background noise reduction, and an attractive design. 3)Panasonic - 8-Microphone Speakerphone with Caller ID KX-TS730S Why it's a best pick: With a 360 10' radius and 8 microphones, everyone is sure to be heard with the Panasonic KX-TS730S. The multiple microphones allows for everyone sitting in on the conference to be heard uniformly without distortion. 4)Cisco Unified IP Conference Station 7937G Conference VoIP Phone Why it's a best pick: The Cisco 7937G works via VoIP connection, has stunning call clarity, and features a simplistic but expensive design that is easy to use. Cisco is an industry leader in IT communication products, and the 7937G is no different. The 360 design allows everyone to be heard. 5)Polycom SoundStation VTX 1000 Why it's a best pick: The SoundStation VTX 1000 is an incredible conference phone, but it is very pricey and not as good as advertised. The VTX 1000 is designed for large conference rooms and features upgradable software (which is a huge benefit since the cost is so high), 20' 360 radius. 6)PolycomR SoundStationR IP 5000 7) GXP2120 6-line Executive HD IP Phone On Wed, Nov 30, 2011 at 1:47 PM, Faisal Hanif fai...@vopium.com wrote: I have tried EyeBeam and it worked fine with x members audio conference however it need resources (Processing + RAM) per additional line. Regards, Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Wednesday, November 30, 2011 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Danny Nicholas; Sam Govind Subject: [asterisk-users] Best VoIP conferencing phone ? Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 tel:%2B91-8885268942 Software Engineer -- _ -- 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 -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- 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
Re: [asterisk-users] Outbound Dial
U can also use VICIDIAL for it -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan Sent: Saturday, August 20, 2011 12:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Outbound Dial Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). is there a utility available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide Regards Kaushal -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Playback while dialing out
Well as far as I know asterisk you can't play anything while channel is in dialing state but music-on-hold. A solution to your problem is realtime music-on-hold. Following are possible steps, 1-Configure your asterisk for realtime music-on-hold (http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf) so it will get all mog class info from DB in realtime. 2-Before dialing a call create a moh class in db by hitting a query and associate your target voice.mp3 files with that class. 3-Dial the call and associate that moh class using parameter. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin Sent: Saturday, August 20, 2011 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Playback while dialing out I am not sure you even read my mail, no music on hold option - it should work dynamically with any file. On Fri, Aug 19, 2011 at 6:18 PM, bakko asannu...@gmail.com wrote: Hi, you can configure a new music on hold, example: nano /etc/asterisk/musiconhold.conf [default1] mode=files directory=moh1 and put the audio file in this directory; then change your dialplan like: exten = 500,1,NoOp exten = 500,2,Dial(SIP/14085551234@myprovider,m(default1)) exten = 503,3,Hangup Regards -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Asterisk+internal phones+recorded messages
You can have all this plus a lot more. What you need is configurations and dialplan code. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of neo haux Sent: Thursday, August 11, 2011 6:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk+internal phones+recorded messages Hi I want to change my old answering phone machine and two wireless phones with asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel 9133i) + Wifi/SIP phone I am wondering if I´ll lost actual functionalities that are present in my old answering machine: 1) is it possible to show the caller number (coming from PSTN/FXO) in both SIP phones (wifi/SIP and LAN phone) ? Does SIP protocol take in charge this functionality 2) Most important question is : can I see on those internal phones (Wifi/SIP phone and LAN phone) that I´ve some recoded messages on asterisk. Indeed, I have this fucntionality with my old answering machine where I can see the number of new messages recorded in a big LCD screen. Thx -- _ -- 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
Re: [asterisk-users] Firewall Issue
If you take a bit deep analyses on SIP packet you will be able to understand the issue, Iptables filter on layer-3 while SIP is on layer-7. It is easily possible to generate a SIP packet with different source-ip than physical interface. You can also simulate it if you set external-ip=some-else-ip in SIP.com in asterisk. All you SIP packets will contain new some-else-ip while layer-3 headers will still have actual physical interface IP. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai Sent: Monday, August 08, 2011 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Firewall Issue On Mon, Aug 8, 2011 at 5:09 PM, Henrik sing...@common-hacking.org wrote: Also you can set allowguest=no in sip.conf, if you didn't do it already I will check sip.conf, but logically, the packets should not be reaching Asterisk. IP Tables should have blocked them. Sans -- _ -- 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
Re: [asterisk-users] dundi
Dundi just give you location of extensions. For ring you should have capable dialplan and peering. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali Sent: Wednesday, August 03, 2011 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dundi Dear is it possible to send ring(call) to all devices with same (sip_username) in all servers ? in this schematics, some bodies have shared lines. so all lines must be in service . Best -- Pezhman Lali -- _ -- 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
Re: [asterisk-users] Why no traction for Windows version?
Hi, I haven't write any How to on it but below are some step by step instructions to run Asterisk on windows, 1-Install Cygwin. 2-Install build essentials in Cygwin. 3-Download Asterisk source (I used 1.4.x) and unzip it using tar (You may need to install tar manually as it is missing in some Cygwin default installations. Don't use windows unzip for as it will create some abnormal character in source and will make unexpected compile time errors) 4-Run bootstrap it will report any missing or lower version libs, prerequisite or tools. 5-You may need to manually install/upgrade tools like autoconf, automake etc depending on your Cygwin installation. 6-You manually need to download and compile termcap, ncurses. 7-Run configure. 8-Make menuselect and disable all non-required modules as it will save to resolve lot of not needed dependencies. 9-Run make 10-Resolve any missing reported by make. 11-After successful make run make install 12-Once make install okey you can run asterisk on Cygwin console and also directly run by double clicking on asterisk.exe in c:/Cygwin/usr/sbin/. Once you have compiled it you can copy asterisk.exe to any other system not having Cygwin installed by you have to care about following, 1-You must have to create required directories structure like Cygwin on system drive. 2-You must need to copy required Cygwin DLLs to new systems \windows\system32\ folder. You can identify required DLLs by trying to run asterisk.exe and it will report missing DLLs one by one. I did just for my experiment and fun and was able to make successful SIP calls using static files configuration. However I suggest to use SIPx, Yate or FreeSWITCH if you want to stick with windows as that have native windows ports and have all required features you need in a PABX or VoIP switch. Regards, Faisal Hanif -- _ -- 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
Re: [asterisk-users] Questions about FMFM with linked servers
Did you tried to execute Set(CALLERID(num)=you-required-callerid)? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman Sent: Friday, July 29, 2011 1:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Questions about FMFM with linked servers All; In a linked server environment, running Asterisk 1.6 I am noticing that when a call is placed from server A to server B (via 4 digit extension) and server B ext has a FMFM to call their mobile, the mobile phone shows the default caller ID setting on server B instead of the actual caller id of the person who initiated the call on server A. This scenario, of course, works in the event a call in placed via the PSTN into Server A (or B) and rings the FMFM extension. In this case, the mobile phones sees the correct (initial) caller ID on the mobile. Thanks! --Dovey -- _ -- 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
Re: [asterisk-users] Why no traction for Windows version?
Hi, One more thing previously there was a project named as AstWin which was maintaining asterisk's port to windows and providing an installable package of Asterisk for windows. I am not aware about current state of project but, I have installation package of Asterisk for windows version 1.2. If anyone need it contact me direct at email imfa...@gmail.com I will send the software as attachment. Regards, Faisal Hanif -- _ -- 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
Re: [asterisk-users] Accept the dtmf input in call patch
Yep. Look the dtails of option of Dial command and features.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vinod Dharashive Sent: Friday, July 29, 2011 8:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Accept the dtmf input in call patch Hi team, Is it possible to capture dtmf input once call is patched between a-party and b-party? Also on dtmf input issue hangup request to b-party with out disconnecting A-party. How is this scenario implemented in dialplan? Thanks Vinod Dharashive Sent from BlackBerryR on Airtel -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Why no traction for Windows version?
I have tried asterisk on windows XP using Cygwin and it worked fine. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antonio Modesto Sent: Thursday, July 28, 2011 1:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why no traction for Windows version? On Tue, 2011-07-26 at 09:45 +0200, Gilles wrote: On Tue, 26 Jul 2011 07:28:27 +, Soeren Malchow (MCon) soeren.malc...@mcon.net wrote: And asterisk just runs fine on linux why bother ? Because I, for one, would like to run Asterisk on my Windows workstation at home as an enhanced answering machine :-) Windows never was a good solution for these things, and i think it will never be. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Multiple SIP trunks between same pair of asterisk box
If it is just matter of billing you can pass billing related info in additional SIP headers on single trunk. If you must need multiple trunk you can add multiple IPs of different subnet class to both interfaces and configure asterisk to listen of all IPs. Then use one trunk per IP Subnet class. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Thursday, July 21, 2011 3:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Multiple SIP trunks between same pair of asterisk box Hello, for billing purpose between a multitenant asterisk box and another asterisk, I am in the need to maintain multiple SIP trunks between them. Usually I use insecure=invite,port but I had to remove or the trunks will be selected based on IP address and not with username/password. I had to use the fromuser option or asterisk will try to authenticate the call using the CID and not the username, but this break the outbound CID of the client. Both are asterisk 1.6 Is there any other solution from multiple SIP trunks between two asterisk boxes? Leandro -- _ -- 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
Re: [asterisk-users] Multiple Asterisk Sessions on same machine
Two very simple solution for your problem: 1-Port redirection in iptables. This I have used for a year or plus and it worked fine for me. I have redirected 1000 ports to a single port 5060 in iptables and it worked smooth. 2-There is a script in asterisk source directory to compile portable asterisk. You can compile asterisk as portable and copy compiled asterisk to multiple locations/directories (as many instances you need). Each copy will have its own configuration files where you can play as you like. Regards, Faisal Hanif -- _ -- 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
Re: [asterisk-users] Problem in Detecting Dtmf on FXO line.
Did u tried by disabling relaxdtmf? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Friday, July 08, 2011 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem in Detecting Dtmf on FXO line. Hi All, I am having Problem in detecting DTMF on analog lines. basically are system is in india and telco provider is BSNL [Bharat sanchar Nigam LImited]. We have Purchased Analog card From chinaroby.com which is X1600P 16 port FXO card. they also provide us wctdm.c file. card is detected successfully, incoming and outgoing calls scenario is also fine. we are unable to receive dtmf properly it means there is some digit are missing when we receive dtmf the ratio of sucess is about to 70% and 30% of calls are getting wrong dtmf . Dahdi version is 2.3.0.1 and asterisk version 1.6.0.24 I load module using modprobe wctdm opermod=INDIA cidbeforering=1 cidbuflen=1 fixedtimepolarity=16 here id chan_dahdi.conf. [trunkgroups] [channels] context=from-zaptel signalling=fxs_ks busydetect=yes busycount=4 ;rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes callerid=asreceived cidstart=polarity_in cidsignalling=dtmf hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callprogess=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 ;cid_rxgain=5.0 relaxdtmf=yes callgroup=1 pickupgroup=1 toneduration=500 ;answeronpolarityswitch=yes hanguponpolarityswitch=yes ;polarityonanswerdelay=1000 group=0 channel = 1 ;channel = 2 ;channel = 3 ;channel = 4 ;channel = 5 ;channel = 6 ;channel = 7 ;channel = 8 ;channel = 9 ;channel = 10 ;channel = 11 ;channel = 12 ;channel = 13 ;channel = 14 ;channel = 15 ;channel = 16 Also set tonezone = in in system.conf, tried many solutions and changed so many parameters of chan_dahdi.cong but still i am not getting successful result. Please share your comments if anyone have idea for india specific region . Regards Dhaval -- _ -- 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
Re: [asterisk-users] dialout time configuration
I think yes. Check queuetimout variable. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deka, Rajib IN MAA SL Sent: Friday, July 08, 2011 3:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dialout time configuration Hi List, Is it possible to configure an infinite ring timeout for queue in asterisk? I mean, the caller should be able to be in queue until and unless he disconnects the call. Thanks, Rajib _ Important notice: This e-mail and any attachment there to contains corporate proprietary information. If you have received it by mistake, please notify us immediately by reply e-mail and delete this e-mail and its attachments from your system. Thank You. -- _ -- 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
Re: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
Use Filter command in dia-plan to get numeric only string,
Set(MYNEWCLI=${FILTER(0123456789,${CALLERID(number)})
Regards,
Faisal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, July 07, 2011 9:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Stripping characters from ${CALLERID(num)} ?
Greetings-
On occasion, I have calls coming into an Asterisk 1.2.x system where the
${CALLERID(num)} includes '-'. Ex:
123-456-7890
How can I strip the dashes from the number, leaving me with '1234567890'?
I've tried the following which does not appear to be working:
Dialplan:
exten = _X.,n,Set(PROPERCID=System(echo ${CALLERID(num)} | sed s/\-//g))
exten = _X.,n,NoOp(Fixed proper CID is ${PROPERCID}
Console Output:
-- Executing [11@cidmangletest:4] Set(SIP/w.x.y.z-b4d55ce8,
PROPERCID=System(echo 123-456-7890 | sed s/\-//g))
-- Executing [11@cidmangletest:5] NoOp(SIP/w.x.y.z-b4d55ce8,
Fixed proper CID is System(echo 123-456-7890 | sed s/-//g))
Obviously, I'm trying to throw the CID through sed via System() to strip the
dashes. Can anyone explain how to accomplish this? Or even better yet, how
to strip the dashes directly in the dialplan without the use of System()?
Thanks!
--Tim
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Re: [asterisk-users] Eyebeam crashes when dialing an invalid number...
As asterisk is an B2BUA you can handle 503 at asterisk and hang caller end using the response code compatible with eyebeam as Hangup(16) Regards, Faisal -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Thursday, July 07, 2011 10:17 PM To: Asterisk Subject: [asterisk-users] Eyebeam crashes when dialing an invalid number... Lately I have been getting many complains that Eyebeam crashes when you dial a number that does not exist. This happens in both R2 and ISDN PRI lines. The softphone stops working and has to be restarted. The response I got from tech support was: the actual issue is that asterisk should not be sending a 503 service unavailable when a particular softphone is not online. The soft phone stops because a 503 means that the server itself is unavailable. Does anyone have a workaround for this? Maybe a way to manipulate via dialplan so the softphone does not get the 503 message? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
Re: [asterisk-users] single keypress short-circuits to invalid extension handler
You can't use WaitExten to receive two digits. Use Read() command. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, July 06, 2011 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] single keypress short-circuits to invalid extension handler Hello all I'm running Asterisk 1.8.4.4 in a new installation. I'm seeing peculiar behaviour in a context where I dispatch to different MeetMe conference rooms. It seems the first digit is being given to Asterisk and it ALWAYS jumps to the i extension. I originally had single digits for the MeetMe rooms, I tried double digits to no avail. As soon as I press the 0 key it plays the invalid message. Here is my meet-me context from my dialplan. Any ideas? Other sections of my dialplan work fine in permitting multiple digit keypresses. I have used this same dialplan in many other installations, so I'm pretty flummoxed by this. Cassius Smith [meet-me] exten = s,1(top),NoOp() same = n,Answer() same = n,Wait(1.0) same = n,Background(enter-conf-call-numberdigits/0digits/0throughdigits/0digit s/9) same = n,WaitExten(5) exten = 00,n,MeetMe(SouthAfrica0,dMs) exten = 01,n,MeetMe(Swaziland1,dMs) exten = 02,n,MeetMe(Botswana2,dMs) exten = 03,n,MeetMe(Zimbabwe3,dMs) exten = 04,n,MeetMe(Lesotho4,dMs) exten = 05,n,MeetMe(Mozambique5,dMs) exten = 06,n,MeetMe(Zimbabwe6,dMs) exten = 07,n,MeetMe(Namibia7,dMs) exten = 08,n,MeetMe(Angola8,dMs) exten = 09,n,MeetMe(Congo9,dMs) exten = t,1,Goto(s,top) exten = i,1,Playback(invalid) same = n,Goto(s,top) And here is the console output. -- Executing [4098@users:1] Goto(SIP/4099-0026, meet-me,s,1) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:2] Answer(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:3] Wait(SIP/4099-0026, 1.0) in new stack -- Executing [s@meet-me:4] BackGround(SIP/4099-0026, enter-conf-call-numberdigits/0digits/0throughdigits/0digits/9) in new stack -- SIP/4099-0026 Playing 'enter-conf-call-number.ulaw' (language 'en_ZA') -- Invalid extension '0' in context 'meet-me' on SIP/4099-0026 == CDR updated on SIP/4099-0026 -- Executing [i@meet-me:1] Playback(SIP/4099-0026, invalid) in new stack -- SIP/4099-0026 Playing 'invalid.slin' (language 'en_ZA') -- Executing [i@meet-me:2] Goto(SIP/4099-0026, s,top) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- _ -- 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
Re: [asterisk-users] Monitoring connection to VoIP provider?
Community can help you better if you provide some details about you scenario and requirement. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Wednesday, July 06, 2011 5:03 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Monitoring connection to VoIP provider? Hello I was wondering if Asterisk can be configured to monitor a connection to a VoIP provider, whether someone is currently using it for a call or the connection is idle? FWIW, my VoIP provider doesn't run an iperf server on their side. I don't know if ping/traceroute is a good enough solution to monitor an SIP connection. I'd like this so I can check how good the line is before calling or receiving a call. Thank you. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] ooh323 does not work fine, what about h323 channel
Hi, As per my experience YATE is the best option for H323=SIP Proxy. Regards, Faisal -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, July 07, 2011 2:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ooh323 does not work fine, what about h323 channel Hi All; The asterisk version I am using is 1.8.4.2 and I compiled ooh323 channel (by selecting the add-on). But really does not work in good performance, for example: if a call came from gnugk to asterisk and the ooh323 handled it, the performance is bad .. some calls are drop and if it is ringing, then it rings for small duration and then stop ringing In other words, if the call went from gnugk to the provider directly (all the path h323), it is better than coming for Asterisk via the ooh323 channel and then to be translated for SIP to be sent for provider. I would like to try the h323 channel (and not the ooh323), but I do not know what I have to do to compile? Any advise? Did anyone tried yate to do the translation from h323 to sip? How it is? Regards Bilal -- _ -- 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 -- _ -- 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
Re: [asterisk-users] realm question
The problem you are reporting is not related to realm but can be context or domain. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Tuesday, July 05, 2011 11:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: j.witvl...@mindef.nl Subject: [asterisk-users] realm question Hi all, Trying to find where i got wrong in my config Is the realm parameter in sip.conf only used for possible autentication? The thing is, i got my box more-or-less working as i wanted, but i can only reach internal functions (like echo-test and so on) and other sip-clients if i dial 1234@fqdn, while i was expected to be able to just dial 1234 I presume i have either a mismatch between how the softphones register, and my asterisk conf. Kind regards, Hans -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Load Balance Trunks
Hi, One of my college Gohar Ahmed suggested an intelligent solution to your problem. I am coping his words below, Create SIP trunks and create a queue [distributor] and register trunks in it as static agents with strategy rrmemory , To keep track of number of calls served per trunk as well as time on each trunk can be monitored via any queue monitoring tool. !! or better use queue_log in realtime DB As per my view this is most easy and optimized approach while keeping all possible data in queue logs. Hope this will helpful for you. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Saturday, July 02, 2011 1:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Load Balance Trunks On Fri, 1 Jul 2011, Abid Saleem wrote: The intention is to load balance between 100 or even more trunks. Filling up one trunk may have another problem because we have another restriction on 5 simultaneous calls per trunk. Yes unused capacity can be rolled over to the next day. Anything is fine that does not break these two restrictions of 120 mins/day/trunk and 5 simultaneous calls/trunk. Please help me in writing an AGI script or whatever required if you can as I am not a programmer. If you don't consider yourself a 'programmer' then you don't have the skills to start. You should hire a competent programmer. It will be much cheaper in the long run and you can focus on what you are good at instead of what you are not. It's not that the requirements are all that challenging, it's just that the probability of success when you lack the skills is small. These skills include, but are not limited to: 1) An understanding of Asterisk, dialplan logic, and applications. 2) An understanding of the AGI interface including reading and setting channel variables. 3) MySQL programming and administration skills. 4) The ability and experience to write well thought out, clearly presented, robust and maintainable code. What service are you offering? Are the calls delivered by SIP or PSTN? Is this a 24x7 operation? If I was asked to design a 500 simultaneous call system with SIP delivery I would probably start with 2 OpenSIPS servers, 2 Asterisk instances (possibly on the same servers as the OpenSIPS servers), and at least 1 MySQL server. You could cram everything on to a single system, I just don't like to put all my eggs in a single basket. I like 'front-ending' Asterisk servers with OpenSIPS because it gives me the flexibility to handle a host failure or take a host out of production for maintenance. AJS (previous poster) has the right approach -- 2 AGIs. One AGI to determine which trunk to use (I would use a 'select' to determine which trunk instead of 'random') and one executed at the end of the call to update the database. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- _ -- 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
Re: [asterisk-users] Couldn't call Agent and segfault
If the problem always related to some specific module then try clean recompiling asterisk if it is with random modules then check you system RAM. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Agustina Berretta Sent: Wednesday, July 06, 2011 1:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Couldn't call Agent and segfault Hi folks! I´m having the following problem: I get the following messages, asterisk get automatically reloaded and agents log out once or twice a day, randomly. [Jul 4 11:36:25] VERBOSE[30004] app_queue.c: -- Couldn't call Agent/2002 [Jul 4 11:36:29] VERBOSE[30320] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log [Jul 4 11:36:29] VERBOSE[30320] config.c: == Parsing '/etc/asterisk/asterisk.conf': [Jul 4 11:36:29] VERBOSE[30320] config.c: == Found [Jul 4 11:36:29] VERBOSE[30320] loader.c: Asterisk Dynamic Loader Starting: [Jul 4 11:36:29] VERBOSE[30320] config.c: == Parsing '/etc/asterisk/modules.conf': [Jul 4 11:36:29] VERBOSE[30320] config.c: == Found [Jul 4 11:36:29] NOTICE[30320] loader.c: 2 modules will be loaded. Also I get a segfault in /var/log/messages. Any help will be appreciated! -- _ -- 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
Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system
You have to provide channel ID to command like channel request hangup SIP/12316156-sad4d46a5. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail] Sent: Wednesday, July 06, 2011 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on Debian / Sparc taking up 95%+ CPU with No calls on the system On Wed, Jul 6, 2011 at 12:00 AM, Daniel - Asterisk earohua...@gmail.com wrote: On the CLI write: sip show channels If there are lots of bye channels you have the same problem than me. I've tried waiting with the call generator -sipp- and channels finished when there are a few. But they're not ending faster enough when I send lots of concurrent calls. Elder Hi, thanks for the response. yeah I'd checked that before and I only have 2 dialogs which seem to be part of the same call that are just sitting there and I can't seem to get them to hang up by typing channel request hangup all . I even tried sending a Hangup by connecting on the AMI but that doesn't seem to be doing anything either. So this channel is sitting there in the 'BYE' state. Is there anyway of clearing them without having to reload/restart Asterisk? I want to see if that's the cause of the CPU usage and I'll lose that if I restart Asterisk. Thanks 2011/7/5, A E [Gmail] all.efor...@gmail.com: hello people, I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some reason I have noticed that only after a few test calls, the asterisk process is running between 95% - 99.9% CPU when there's absolutely nothing on the system. This is a clean Asterisk system in an internal network with nothing else on it with no calls on it but it's still sitting with 96% CPU. I'm not a developer so not that ept with using debug tools etc to figure out why it's doing that. Could anyone please tell me how I can figure out why it's doing this and/or help debug this. Makes no sense for it to be using CPU with nothing happening on the system Thanks -- Enviado desde mi dispositivo móvil -- _ -- 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 -- _ -- 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
Re: [asterisk-users] how to set to make a call through a fixed ip on a 2 ips server?
Hi, I don't think there is a way for it inside asterisk but you achieve it by adding static route in Linux routing table and make interface having that IP as default route for the interested IPs traffic. Regards, Faisal From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cnasterisk Sent: Monday, July 04, 2011 10:40 AM To: asterisk-users Subject: [asterisk-users] how to set to make a call through a fixed ip on a 2 ips server? Hi all, I have a server runing asterisk 1.8, and the server has 2 different ip address if i want to make a call from a sip trunk with a fixed ip from the 2 ips, how to do? 2011-07-04 _ cnasterisk -- _ -- 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
Re: [asterisk-users] SIP Peer Name Variable
When you make a call asterisk always create a channel named as below,
CheannelType/PeerName-uniquecode
Like
SIP/jon-312abf
So here jon is the peer name. This can help you to identify a peer as long
as A-Leg is active.
Regards,
Faisal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Sunday, July 03, 2011 6:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Peer Name Variable
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan
Journo
Sent: Saturday, July 02, 2011 8:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP Peer Name Variable
Hi,
Is there a variable that contains the Sip Peer name?
I was using ${CALLERID(num)} for outgoing calls, but when a call is
being transferred, that variable contains something else.
I need a variable that is always set to the SIP Peer's name.
pbx*CLI core show function CHANNEL
-= Info about function 'CHANNEL' =-
[Synopsis]
Gets/sets various pieces of information about the channel.
[Description]
Gets/sets various pieces of information about the channel, additional item
may be available from the channel driver; see its documentation for details.
Any item requested that is not available on the current channel will
return an empty string.
[Syntax]
CHANNEL(item)
[Arguments]
item
Standard items (provided by all channel technologies) are:
audioreadformat - R/O format currently being read.
audionativeformat - R/O format used natively for audio.
audiowriteformat - R/O format currently being written.
callgroup - R/W call groups for call pickup.
channeltype - R/O technology used for channel.
checkhangup - R/O Whether the channel is hanging up (1/0)
language - R/W language for sounds played.
musicclass - R/W class (from musiconhold.conf) for hold music.
name - The name of the channel
parkinglot - R/W parkinglot for parking.
rxgain - R/W set rxgain level on channel drivers that support it.
secure_bridge_signaling - Whether or not channels bridged to this
channel require secure signaling
secure_bridge_media - Whether or not channels bridged to this channel
require secure media
state - R/O state for channel
tonezone - R/W zone for indications played
transfercapability - R/W ISDN Transfer Capability, one of:
SPEECH
DIGITAL
RESTRICTED_DIGITAL
3K1AUDIO
DIGITAL_W_TONES
VIDEO
txgain - R/W set txgain level on channel drivers that support it.
videonativeformat - R/O format used natively for video
trace - R/W whether or not context tracing is enabled, only available
*if CHANNEL_TRACE is defined*.
*chan_sip* provides the following additional options:
peerip - R/O Get the IP address of the peer.
recvip - R/O Get the source IP address of the peer.
from - R/O Get the URI from the From: header.
uri - R/O Get the URI from the Contact: header.
useragent - R/O Get the useragent.
peername - R/O Get the name of the peer.
t38passthrough - R/O '1' if T38 is offered or enabled in this channel,
otherwise '0'
rtpqos - R/O Get QOS information about the RTP stream
This option takes two additional arguments:
Argument 1:
'audio' Get data about the audio stream
'video' Get data about the video stream
'text' Get data about the text stream
Argument 2:
'local_ssrc'Local SSRC (stream ID)
'local_lostpackets' Local lost packets
'local_jitter' Local calculated jitter
'local_maxjitter' Local calculated jitter (maximum)
'local_minjitter' Local calculated jitter (minimum)
'local_normdevjitter'Local calculated jitter (normal
deviation)
'local_stdevjitter' Local calculated jitter (standard
deviation)
'local_count' Number of received packets
'remote_ssrc' Remote SSRC (stream ID)
'remote_lostpackets'Remote lost packets
'remote_jitter' Remote reported jitter
'remote_maxjitter' Remote calculated jitter (maximum)
'remote_minjitter' Remote calculated jitter (minimum)
'remote_normdevjitter'Remote calculated jitter (normal
deviation)
'remote_stdevjitter'Remote calculated jitter (standard
deviation)
'remote_count' Number of transmitted packets
'rtt' Round trip time
'maxrtt'Round trip time (maximum)
'minrtt'Round trip time (minimum)
'normdevrtt'Round trip time (normal deviation)
'stdevrtt' Round trip time (standard deviation)
'all'
Re: [asterisk-users] Outgoing calls get dropped on high-latency connections.
Have you tried SIP session timer values in sip.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar Sent: Tuesday, June 28, 2011 9:33 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Outgoing calls get dropped on high-latency connections. We're a VoIP provider essentially competing with our local incumbent Telco, and a sizeable number of our customers use satellite internet. As a result, these customers never have ping times less than 500ms, and are often exceeding 2500ms. I manually apply a patch to the Asterisk source code every time we upgrade Asterisk, described here: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg178034.html This change allows our satellite customers to maintain their SIP connection for more than 5 minutes. But we're currently using Asterisk 1.6.2.17, and this version seems to have one very strange bug on these high latency connections. On outgoing and *only* outgoing calls, the call drops after two or three minutes. Incoming calls do not have this problem, so I don't think it's the SIP connection getting killed due to a slow INVITE response. Has anyone heard of this bug? Or should I submit a new bug report to the Asterisk project? This message was sent using Lightspeed.ca's Advanced Webmail. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] not find files in asterisk 1.8
Have you installed sample configuration files package? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paolo De Michele Sent: Monday, June 27, 2011 4:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] not find files in asterisk 1.8 hi list, I have a problem with Asterisk 1.8 I installed the software via the yum repositories of asterisk.org but if I go to the /etc/asterisk/ I do not find any files in it? possible? thanks in advance p -- _ -- 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
Re: [asterisk-users] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI
Call file are not suitable for you as asterisk process these files in serial
mode (single threaded) and in case of large number of files processing of
last file can be that much delayed that some portion of message may be
already played or the 1st phone may be hanged.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Pierce
Sent: Monday, June 27, 2011 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast
with DAHDI
We just finished an upgrade of our Asterisk system to an HA environment on a
pair of servers using Linux-HA. As part of the upgrade, we also moved to
Asterisk version 1.8.4.3
Most things are working quite nicely on the new system. However, I'm having
trouble getting a paging feature to work. In Asterisk 1.4, we simply used
the Page() application like this:
3400,n,Page(SIP/3011SIP/3021SIP/3110SIP/3120SIP/3121SIP/3122SIP/3124S
IP/3125SIP/3126SIP/3127SIP/3221SIP/3222SIP/3223SIP/3250SIP/3261SIP/3
262SIP/3310SIP/3311SIP/3324SIP/3329SIP/3331SIP/3332SIP/3350SIP/3455
SIP/3457)
However, the Page() application seems to rely on the Meetme() application
which also relies on the DAHDI channel driver for mixing of the audio
streams. I have tried using the DAHDI channel driver on this system, but
that seems to make the Music On Hold application use the DAHDI timing module
instead of the pthread module. With the DAHDI timing module, Music On Hold
does not playback Shoutcast streams which is also a requirement for this
system.
As an alternate solution, we have tried implementing a workaround which
simply uses a set of .call files to dial each phone. Those phones then
auto-answer the call and are placed into a conference bridge on mute using
the ConfBridge application. At this point, the initiating caller speaks the
announcement and the phones automatically hangup after about 10 second. This
worked perfectly in our small scale tests. However, when we ramped this up
to the 25 phones that are required and tested it this morning, somehow this
caused the Asterisk service to restart. I suspect that processing the 25
call files and placing them into the conference all at the same time somehow
made the system crash and it immediately started up again.
Here's the relevant dialplan:
exten = 3400,1,Answer
exten = 3400,n,playback(beep)
exten = 3400,n,system(cp /etc/asterisk/testPage/*.call
/var/spool/asterisk/outgoing_staging/)
exten = 3400,n,system(mv /var/spool/asterisk/outgoing_staging/*.call
/var/spool/asterisk/outgoing/)
exten = 3400,n,ConfBridge(testPage,1)
exten = 3400,n,hangup
[testPage]
exten = s,1,Answer
exten = s,n,playback(beep)
exten = s,n,Set(TIMEOUT(absolute)=10)
exten = s,n,ConfBridge(testPage,m)
exten = s,n,hangup
exten = _,1,SIPAddHeader(Alert-Info: Auto Answer) exten =
_,n,Dial(SIP/${EXTEN}) exten = _,n,Hangup()
and here's a sample call file:
channel: Local/3011@testPage
callerid: Page
context: testPage
extension: s
priority: 1
archive: no
waittime: 120
Does anyone have insight into how we could accomplish this paging feature or
of anything that we may have missed?
I suspect we could get this all to work with the original Page() application
if there was a way to force MusicOnHold to use the pthread timing module
instead of the Dahdi timing module. Is that configurable somewhere?
Thanks for your help,
Bob
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Re: [asterisk-users] Conference feature
If you can explain a bit more what exactly you need? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Monday, June 27, 2011 9:16 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Conference feature I am given to understand that it does not. On 06/27/2011 12:13 AM, C F wrote: Does asterisk support it? On Sun, Jun 26, 2011 at 9:25 PM, Rafael dos Santos Saraiva rafaels...@gmail.com wrote: Hi How to create the conference feature in Asterisk? Thank's Att, Rafael Saraiva -- _ -- 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 -- _ -- 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 -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Vm on a System running Asterisk.
It depend on Hypervisor. if it is full virtualization then it will not be more than a part sharing from system resources depends on VM configuration and processing load. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot Sent: Friday, June 24, 2011 12:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Vm on a System running Asterisk. Would it create any problem for Asteisk, if we install Windows as a VM on a system that has CentOS running Asterisk as the base? System also has a PRI card. TYIA, [SATISH] Mumbai, India. -- _ -- 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
Re: [asterisk-users] Monitor Asterisk and Ast-gui
Asterisk-SNMP could be an option for u. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu Sent: Friday, June 24, 2011 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Monitor Asterisk and Ast-gui Hey, I have installed asterisk 1.8 on Slackware 13.1 from source and it is working well. I have 300 ip phones in a natted environment and my asterisk server has a public IP I would love to monitor my SIP activity on my VOIP Server, statistics like amount of sip traffic, who made what call and to whom, how many calls were made in a month, how many ip phones are up and running, which sip phone has made most calls among others. How best can I do that? On the other hand, I have also tried installing ast-gui onto asterisk 1.8, it has installed well but it however keeps looping whenever i try to login in, it says checking permissions on gui folder and loops. Haven't found much help on other mailing lists, any direction given in welcome. Thanks Richard Zulu Twitter www.twitter.com/richardzulu Skype: zulu.richard There is no place like 127.0.0.1 -- _ -- 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
Re: [asterisk-users] How to secure our Asterisk server from hacker's ?
Fail2ban From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Thursday, June 16, 2011 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to secure our Asterisk server from hacker's ? Hi List, Yes you are right but I want to cross check to outside world to. How they will support me in such case... :) On Thu, Jun 16, 2011 at 11:23 AM, Alex Balashov abalas...@evaristesys.com wrote: I thought the idea was that Asterisk Engineers already know the answers to such questions? On 06/16/2011 01:52 AM, virendra bhati wrote: Hi List, I want to secure my server from the hacker's. What is the case by which I can protest it. I have done security of Dialplan, Sip,IAX base security. For linux we are working on Iptables. What else is left so that I will do it too... -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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 -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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
Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
Try by reversing the line number of permit deny -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Thursday, March 10, 2011 6:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL Thanks. But Like I said, that's all done. Here's the Endpoint config: [authentication] [basic-options](!) ; a template dtmfmode=rfc2833 context=Phones type=friend contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 deny=0.0.0.0/0.0.0.0 permit=172.16.16.0/24 host=dynamic qualify=no insecure=port,invite [natted-phone](!,basic-options) ; another template inheriting basic-options nat=yes directmedia=no [555](natted-phone) secret=$$ecret$$ disallow=all allow=ulaw allow=gsm no deal! The irony is that we have a similar configuration at another place, but we didn't need to put anything there and the phones register regardless! Is this broken Perhaps the contactdeny is taking precedence in 1.8. Try it without the contactdeny - maybe the existence of a contactpermit will imply a contactdeny of everything else. Cheers, j -- _ -- 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
Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
One more thing check if your SBC is configured in relay mode or forward mode. If it is in relay mode you will have original SIP-UA IP in all requests coming on asterisk and only SBC IP in via but if it is forward mode you may can have SBC IP all the way in all requests. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Thursday, March 10, 2011 1:42 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL Pay attention, you have permit=172.16.16.0/24 whereas suggestion was permit=0.0.0.0/0.0.0.0 On 3/10/2011 1:48 AM, RR wrote: On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif fai...@vopium.com wrote: You can add following line to your peers configuration permit=0.0.0.0/0.0.0.0 It will allow to use that peer's account from any IP Thanks. But Like I said, that's all done. Here's the Endpoint config: [authentication] [basic-options](!); a template dtmfmode=rfc2833 context=Phones type=friend contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 deny=0.0.0.0/0.0.0.0 permit=172.16.16.0/24 host=dynamic qualify=no insecure=port,invite [natted-phone](!,basic-options) ; another template inheriting basic-options nat=yes directmedia=no [555](natted-phone) secret=$$ecret$$ disallow=all allow=ulaw allow=gsm no deal! The irony is that we have a similar configuration at another place, but we didn't need to put anything there and the phones register regardless! Is this broken -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Is this true for Asterisk as SBC?
Asterisk doesn't have all features of SBC like relay and forward request on packet level but all depends on your scenario what you need. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, March 10, 2011 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Is this true for Asterisk as SBC? Hi All, I have starting to reading About SBC and found one artical reagding SBC and they gives a solutions like this. i want to know is this true in realtime sceanario while we think of an big implementation and is it possible with cloud computing. i have found from http://www.smartvox.co.uk/products_gateways_explained.htm Asterisk as a Session Border Controller Equip the Asterisk server with two ethernet ports, connect one to the Internet and the other to your internal network; set up the firewall, configure the dial plans and you've got everything you need for a fully functional Session Border Controller. * IP phones can register with the SBC either from the internal network or from the Internet. * Use your SBC as an Inbound and/or Outbound proxy to have complete control over incoming and outbound calls * Use it to control access to your IPBX and to overcome the usual problems associated with interfacing VoIP between your private network and the Internet * Solve one-way audio and other notoriously difficult and annoying NAT traversal problems while, at the same time, improving your systems security regards dhaval -- _ -- 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
Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
It just have ACL concept. You can add permitted IPs List to any peer then only from that IPs user can register. If you want to permit all you can add 0.0.0.0 to ACL From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR Sent: Thursday, March 10, 2011 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet -- SBC -- Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13120 parse_register_contact: Domain '172.16.16.6:5060' disallowed by contact ACL (violating IP ) [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13837 register_verify: Registration denied because of contact ACL Note, that the server IP is 172.16.16.11 and the SBC internal Interface IP is 172.16.16.6 the following lines have been added to sip.conf dynamic_exclude_static = yes autodomain=yes domain=172.16.16.6 allowexternaldomains=no In addition, in the general endpoint template in sip.conf, I have the lines contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 host=dynamic What else am I missing? Thanks \RR -- _ -- 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
Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL
You can add following line to your peers configuration permit=0.0.0.0/0.0.0.0 It will allow to use that peer's account from any IP From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR Sent: Thursday, March 10, 2011 11:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL On Thu, Mar 10, 2011 at 12:20 AM, Faisal Hanif fai...@vopium.com wrote: It just have ACL concept. You can add permitted IPs List to any peer then only from that IPs user can register. If you want to permit all you can add 0.0.0.0 to ACL Thanks. but could you be a little more specific? I have added the local net 172.16.16.0/24 almost everywhere I can think of, but it keeps giving that error. Even in sip.conf in the template for company IP phones, I've added contactpermit as well as just permit=172.16.16.0/24 but it still complains about that From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR Sent: Thursday, March 10, 2011 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] [1.8] Unable to Register: Registration denied because of contact ACL Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet -- SBC -- Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]: netsock2.c:94 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13120 parse_register_contact: Domain '172.16.16.6:5060' disallowed by contact ACL (violating IP ) [Mar 10 11:53:59] WARNING[21272]: chan_sip.c:13837 register_verify: Registration denied because of contact ACL Note, that the server IP is 172.16.16.11 and the SBC internal Interface IP is 172.16.16.6 the following lines have been added to sip.conf dynamic_exclude_static = yes autodomain=yes domain=172.16.16.6 allowexternaldomains=no In addition, in the general endpoint template in sip.conf, I have the lines contactdeny=0.0.0.0/0.0.0.0 contactpermit=172.16.16.0/255.255.255.0 host=dynamic What else am I missing? Thanks \RR -- _ -- 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 -- _ -- 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
Re: [asterisk-users] (fast) AGI and AMI synchronization ?
AMI is single threaded link so waiting on it will bring things to hang mode but FastAGI dialplan is multithread. Better to manage all info by AMI in a local hash or array and use sleep/waiting on AGI till required info populated to hash/array by AMI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Corentin Le Gall Sent: Tuesday, March 08, 2011 4:31 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (fast) AGI and AMI synchronization ? Hi, I've been developing some CTI software around asterisk for a while, mainly with the help of AMI and fast AGI. It works quite fine, but I have some trouble sometimes with the un-synchronized property of these 2. Let me explain, we have a dialplan like this one : exten = s,n,UserEvent(useful_input_data) (...) a few actions exten = s,n,AGI(agi://127.0.0.1:/fetch,queuename) The idea is to setup a cti server that talks with both AMI and AGI channels, the first one mainly when one just want to send some data from asterisk to the cti server, and the second one when the dialplan needs some data from this server. My issue is that the AGI requests are received (from the CTI server point of vue) a little bit before the AMI events. In most cases, I don't really care because it is only a little, and the data asterisk needs to fetch from the AGI are set on time. But sometimes not, especially in cases like above, when there are only a few dialplan lines between UserEvent and AGI ... In order to handle that, I thought let's make a sync/meeting point, with the help of the AMI NewExten event, when the app is AGI. The idea would be to keep the AGI connection open as long as the good AMI NewExten event is not received, then to reply and close it, in order for the dialplan to proceed. However, when trying to do this, nothing more occurs on the AMI connection, thus I come to a deadlock ... My question is then, before switching to -dev issues : is there an option somewhere to handle this, whether on the AMI or on the AGI side ? The asterisk version we've been using for a long time is 1.4 and my current attempts are done on 1.8 branch. Thanks, -- Corentin LE GALL Proformatique (Groupe Avencall) - 10bis rue Lucien Voilin - F-92800 Puteaux Tel (+33/0)1.41.38.99.60 - Fax (+33/0)1.41.38.99.70 http://wiki.xivo.fr -- _ -- 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 -- _ -- 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
Re: [asterisk-users] [1.4] Reading phone number the French way?
When you compile asterisk you can select multiple language files by using
make menuselect additionally you find lot of free sources on internet for
language files. Simply create a folder with language short-code in sounds
and then set channel's language variable to that short-code.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Tuesday, March 08, 2011 5:22 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] [1.4] Reading phone number the French way?
Hello,
I need to write a script which prompts the callee to type a number, and then
read it back to them as confirmation:
=== extensions.conf
[robocall]
;Expect 10-digit number excluding final #, 2 tries, 20s time-out exten =
s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20)
exten = s,n,GotoIf($[${LEN(${NBR2CALL})} != 10]?end) ;exten =
s,n,SayDigits(${NBR2CALL}) exten = s,n,SayNumber(${NBR2CALL})
exten = s,n(end),Hangup()
===
Besides the fact that my Asterisk setup only has US sound files in
/var/lib/asterisk/sounds/digits/, I was wondering how to get Asterisk to
read back the number the French way, ie. digits are read by pairs to the
exception of the leading tuple that always starts with 0.
For instance, a landline number in Paris like 01 42 92 81 00 is read
zero-one, forty-two, ninety-two, eighty-one, zero-zero, where I assume
Americans would read all the digits individually (zero, one, four, two,
etc.)
Has someone already looked into this and knows how to solve it?
Thank you.
--
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Re: [asterisk-users] [1.4] Reading phone number the French way?
You can also set it in dialplan using Set(LANGUAGE=FR) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Tuesday, March 08, 2011 5:46 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [1.4] Reading phone number the French way? On Tue, 8 Mar 2011 17:31:26 +0500, Faisal Hanif fai...@vopium.com wrote: When you compile asterisk you can select multiple language files by using make menuselect additionally you find lot of free sources on internet for language files. Simply create a folder with language short-code in sounds and then set channel's language variable to that short-code. Thanks but will using language=fr in zapata.conf be enough to have Asterisk read numbers the right way? -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Loudness of recorded wav-audio
This settings are for ISDN configurations I think. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Monday, March 07, 2011 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudness of recorded wav-audio I tried to ajust the tx- and rxgain for the sip peer in sip.conf. And restarted the asterisk. But it takes no effect. Any suggestion? 2011/3/4 Danny Nicholas da...@debsinc.com Defaults are 0.0 (leave volume unchanged) +values make volume louder, - softer. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Friday, March 04, 2011 8:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudness of recorded wav-audio Could yoz tell me the default value of rxgain or txgain, if there is no rxgain or txgain in conf-data defined? Von meinem iPad gesendet Am 04.03.2011 um 15:34 schrieb Danny Nicholas da...@debsinc.com: In sip.conf, add rxgain=-4.0 to the peer. This (feel free to correct) should reduce the incoming volume by 4 decibels. You’ll have to do a “sip reload” for this to take effect. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Friday, March 04, 2011 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudness of recorded wav-audio Thank you! How can I reduce the RXgain? Am 04.03.2011 um 15:21 schrieb Danny Nicholas da...@debsinc.com: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Friday, March 04, 2011 2:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Loudness of recorded wav-audio Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix two options are: 1. reduce RXgain – assuming your are using Record() command 2. use sox to reduce the volume; something like sox –v .8 file1.wav file2.wav -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Early codec selection / negotiation
If you dialout call without answering and allow all codec for both peers then codec negotiation will be direct between endpoints and asterisk will only do media pass-through. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francois Marier Sent: Sunday, March 06, 2011 7:52 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Early codec selection / negotiation Hi, This seems to be a fairly common question, but I have Googled for this quite a bit and looked at the Asterisk documentation/book and haven't been able to find an answer. My question is: Can I get my IP phone to select a different codec depending on the final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and ulaw) - PSTN Peer(supports g729 and ulaw) - Remote Asterisk Peer (supports speex and ulaw) Currently, it's configured like this: [snom300] disallow=all allow=ulaw [pstnpeer] disallow=all allow=ulaw [asteriskpeer] disallow=all allow=speex which translates to this: Snom300 ---ulaw--- (pass-thru) ---ulaw PSTNPeer Snom300 ---ulaw--- (transcode) ---speex--- AsteriskPeer In other words, my Snom phone always talks to my Asterisk box using the ulaw codec. My Asterisk box then makes PSTN calls using ulaw and Asterisk calls using speex (transcoding in the case of speex). What I'd like to get is this: (1) Snom300 ---g729--- (pass-thru) ---g729 PSTNPeer (2) Snom300 ---ulaw--- (transcode) ---speex--- AsteriskPeer I can get (1) by using this config: [snom300] disallow=all allow=g729 ; only allow g729 [pstnpeer] disallow=all allow=g729 and I can get (2) by using this config: [snom300] disallow=all allow=ulaw ; only allow ulaw [asteriskpeer] disallow=all allow=speex but I can't get both of them to work at the same time since the Snom phone always connects to my Asterisk box using its prefferred codec. If I configure the phone like this: [snom300] disallow=all allow=g729 ; preferred codec allow=ulaw then (2) will fail because it's trying to do this: Snom300 ---g729--- (transcode) ---speex--- AsteriskPeer and it can't transcode g729 to speex without a patent license. If I configure the phone like this: [snom300] disallow=all allow=ulaw ; preferred codec allow=g729 then (1) will fail because it's trying to do this: Snom300 ---ulaw--- (transcode) ---g729 PSTNPeer This is the best description of the problem I've found online: http://fonality.com/trixbox/forums/trixbox-forums/open-discussion/codec-sele ction-negotiation-and-tweaking but unfortunately it doesn't come with a solution. Is there a way to prevent my IP phone from always connecting to my Asterisk box using its preferred codec or is that simply impossible? Cheers, Francois -- Francois Marier identi.ca/fmarier http://feeding.cloud.geek.nz twitter.com/fmarier -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring?
http://www.danielaliaman.com/blog/files/AsteriskSNMPtutorial.pdf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Sunday, March 06, 2011 10:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring? Hi Everyone, I have been searching the web and I don't know if SNMP is just that complex to setup or that not many people use SNMP to monitor Asterisk but the information is scattered all over. I have got to the point to configure SNMP with Asterisk and then it's all confusing from there on to actually see the graphs in Cacti. I would appreciate it if you can post your steps or point me to a good guide posted somewhere on the web. I have followed this but it's not complete: http://www.voipphreak.ca/2008/10/28/asterisk-snmp-with-cacti-howto-upgraded- for-asterisk-16-and-ubuntu/ ***Please don't post any smart-aleck comments like google it. Thanks, -- _ -- 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
Re: [asterisk-users] Gosub and 'h' (again?)
Well a solution for you to put original context name in variable and then
use that variable in goto statement on h.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Friday, March 04, 2011 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Gosub and 'h' (again?)
Problem as follows:
[default]
exten = 777,1,Gosub(sub,1,1)
exten = 777,n,Hangup()
exten = h,1,NoOp(hung up in 'default' context)
[sub]
exten = 1,1,NoOp(in sub)
exten = 1,n,Playback(tt-monkeys)
exten = 1,n,Return()
exten = h,1,NoOp(hung up in 'sub' context)
This works fine if the caller listens to all the 'tt-monkeys' and let's the
system hangup. You get the hang up in the 'default' context.
But, if the caller hangs up BEFORE the end of 'tt-monkeys' - the hang up
occurs in the 'sub' context. This means that I have to force each sub
routine to go to the main contexts 'h' extension ('exten =
h,1,Goto(default,h,1)' in this case).
Is there a way to tell * to use the default 'h' extension on a hang up -
rather than having to put a 'h' in to every separate sub routine?
I know Tilghman said ...Gosub, on the other hand, isn't really even
executing at that point, so there isn't a code path that exists whereby the
Gosub can empty the return stack and return to the original place [see
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033153.html].
But what does that mean in English ;)?
Thanks
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Re: [asterisk-users] GXW4004 - lines get stuck
1-Check signaling type on gateway PSTN ports 2-Set RTP timeout in SIP trunk. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, March 04, 2011 7:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] GXW4004 - lines get stuck Hi, I have an issue with a GWX4004 used a as a VoIP trunk to PSTN lines converter. In some instances, lines get stuck (both parties hang up, but the GXW4004 status shows off hook for the lines). It stays like this until reboot. Is there a specific setting I should be looking for? I couldn't find anything about that specifically. Mike -- _ -- 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
Re: [asterisk-users] Help Asterisk / API / Perl
AstPP jbilling
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Saturday, March 05, 2011 10:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help Asterisk / API / Perl
On Sat, 5 Mar 2011, Olivier CALVANO wrote:
i want use the API on my asterisk 1.6, but i have a small problems :
$typ don't have SIP or IAX, same test without succes:
$typ = $AGI-get_variable('type');
'agi_type' is part of the AGI environment, not a channel variable.
Read the documentation for your AGI library to see how to access the AGI
environment variables -- the cruft Asterisk writes to the STDIN of your AGI
before any of your requests.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] VoIP Bandwidth Calculator
You can find lots by googling but none can give realtime stats as it depends on network. Packet drop, retransmit, codec type will make lot of vibrations From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Thursday, March 03, 2011 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VoIP Bandwidth Calculator Hi, Does anyone have a good VoIP Bandwidth Calculator? Thanks Dan Journo Kesher Communications (UK) Business Phone Systems http://www.keshercommunications.com/ | Hosted PBX http://www.keshercommunications.com/hostedpbx.html -- _ -- 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
Re: [asterisk-users] How do I find a phone numbers issued by Rogers?
I don't remember exact name but there are two authorities which provide real-time portability information online but you need subscription. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Thursday, March 03, 2011 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How do I find a phone numbers issued by Rogers? On Wednesday 02 March 2011 19:17:03 Robert Augustyn wrote: Is there a way of finding out what block of phone numbers were issued to Roger’s business customers in my end of the woods? You can find out from NANPA, the registry which assigns blocks of phone numbers. Note that due to phone number portability, however, this only will tell you the numbers that were originally allocated to Rogers, as customers are free to request existing numbers to be ported to them, and former customers are free to port their numbers away from Rogers. -- Tilghman -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Testing from where number is...
www.numberingplans.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski Sent: Thursday, March 03, 2011 12:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Testing from where number is... Hi! My customer want's to allow calls to landlines in EU and US and disallow calls to cells in EU. Rest of countries are blocked. Country blocking is easy... Is there a service that allows checking phone number? Maybe some specific Enum? I ask for number and server responds with info, for example: Cell Phone, Belgium or Land Line, Germany. -- Piotr Gorski -- _ -- 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 -- _ -- 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
Re: [asterisk-users] two questions regarding incoming call
You don't need to put quotes around AGI name.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan Kayhan
Sent: Tuesday, March 01, 2011 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] two questions regarding incoming call
Hello,
I want to make an agi script to match incoming DIDs with usernames.
I tried to do such entry in incoming trunk.
[DID_diddw]
include = from-didww
[from-didww]
exten = 3130XXX,1,AGI(did.php)
exten = 3130XXX,n,DIAL(SIP/${yup_no},20)
but when i run the rule it says
chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to
extension '3130111' rejected because extension not found in context
'from-didww'
Cant I use such agi scripts on incoming calls?
PS:
exten = 3130XXX,n,DIAL(SIP/) works alone.
My second question.
I got two incoming trunk sip channels on my server.
One of them is as follows.
[46.19.209.1]
host = 46.19.209.1
type = friend
insecure = invite
context = from-didww
canreinvite=no
The other is as follows:
[62.180.237.73]
host = 62.180.237.73
type = friend
insecure = invite
context = from-btnet2
canreinvite = no
The problem is, i get all calls coming from trunk1(didww) without a problem
but, when i receive a call from trunk2(btnet) it tries to authenticate the
sip call and denies it. It works only if i allow guest calls.
What can be the reason for that?
Thank you.
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Re: [asterisk-users] DIAL through Specific number in PRI
If your PRI provider permit you to associate any ANI to any Circuit-ID you can do this. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, February 24, 2011 12:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DIAL through Specific number in PRI Hi ALL, I have PRI line everything is fine , but my customer having a requirement that they want to DIAL a number from PRI which gives callerid as Specific number. i.e PRI start from 3055 to 30550100 i have purchased a 100 number from telco and our pilot number is 3055, now if some caller want to dial any number but caller should shown is 30550008 like this. is there any solution from asterisk side. regards Dhaval -- _ -- 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
Re: [asterisk-users] DTMF and Snom
Well you simple use dtmfmode=info in peer configuration of Snome phone. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Friday, February 18, 2011 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DTMF and Snom Hello list, I'm having some troubles with DTMF tones. When pressing numbers on a Snom phone, the DTMF-signal takes too long. I have the following in sip.conf : dtmfmode = rfc2833 which works well for Grandstream, Yealink and Cisco phones. But not for Snom. Snom support tells me I should use SIP info. Is it possible to have something like this : dtmfmode = rfc2833, info ?? Because all the other phones types are set to rfc2833, I cannot change to just dtmfmode = info What is a proper solution in my case here ?! Thank you ! Jonas. -- _ -- 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
Re: [asterisk-users] Dial(Local/...) vs. Goto()?
The difference you will feel when using callback files or AMI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Friday, February 18, 2011 1:31 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dial(Local/...) vs. Goto()? Hello, I was wondering: What does Dial(Local/...) offer that a Goto() doesn't? For instance: ;exten = h,n,Goto(callback,start) exten = h,n,Dial(Local/start@callback) [callback] exten = start,1,Verbose(In callback) Thank you. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Asterisk with TE 121 DADHI incoming calls fail
This is not Digium's customer support address but free public emailing list
for asterisk user's contributed by community volunteers.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jan Zieher
Sent: Friday, February 18, 2011 2:19 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk with TE 121 DADHI incoming calls fail
Dear Customer Support,
i connected the asterisk to a e1 interface of our hipath4000. outgoing calls
from a sip peer of my asterisk to an up0 telephone which iss connected to
the hipath4000 are working. If you want to dial from an up0 device to the e1
interface where asterisk is connected to, you have to use the prefix 83. But
when you enter the 3rd cipher this error appears at the cli
CODE:
http://forums.digium.com/viewtopic.php?f=1t=77155sid=3679c2a13cbbf9aa2e75
d02a2f00c8b3 SELECT ALL
[Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1 !! Not
yet handling pre-handle message type SEGMENT (0x60)
[Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1 !!
Don't know how to pre-handle message type SEGMENT (0x60)
[Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1 !! Not
yet handling pre-handle message type SEGMENT (0x60)
[Feb 16 11:29:07] ERROR[2555]: chan_dahdi.c:13691 dahdi_pri_error: 1 !!
Don't know how to pre-handle message type SEGMENT (0x60)
My Config files:
extensions.conf
CODE:
http://forums.digium.com/viewtopic.php?f=1t=77155sid=3679c2a13cbbf9aa2e75
d02a2f00c8b3 SELECT ALL
[general]
static=yes
writeprotect=no
[isdn]
; Ankommende anrufe
exten = 833762,1,Dial(SIP/3762,45,r)
; Rausgehende Anrufe
exten = _0[1-9].,1,Dial(DAHDI/g1/${EXTEN:1})
[default]
include = isdn
/etc/dahdi/system.conf
CODE:
http://forums.digium.com/viewtopic.php?f=1t=77155sid=3679c2a13cbbf9aa2e75
d02a2f00c8b3 SELECT ALL
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
# Global data
loadzone= de
defaultzone = de
chan_dahdi.conf
CODE:
http://forums.digium.com/viewtopic.php?f=1t=77155sid=3679c2a13cbbf9aa2e75
d02a2f00c8b3 SELECT ALL
[trunkgroups]
[channels]
language=de
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
internationalprefix = 00
nationalprefix = 0
;localprefix = VORWAHL
;privateprefix = VORWAHL+MSN
;unknownprefix =
priindication = outofband
facilityenable = yes
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
immediate=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
callgroup=1
pickupgroup=1
mohinterpret=default
mohsuggest=default
overlapdial=yes
group=1
signalling = pri_cpe
channel = 1-15,17-31
context = default
I would be gratefully, if you have an idea or some advices to me.
Thanks !
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Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2
Did you checked if you extension.ael doesn't have syntax error? Did you upgraded anything after last compile? Or Try a clean recompile Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin Sent: Friday, February 18, 2011 4:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2 Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module 'pbx_ael.so' could not be loaded. I did not find in google what it could be and what should be done to solve this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind debian as OS and install asterisk from sources that I took on digium site. Did anyone have the same issue? Regards, Kate -- _ -- 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
Re: [asterisk-users] lua -asterisk manual
The only specific you need to do in extensions.lua is create a table to put
your extensions in like,
Extension{
}
Else all will be LUA code and all asterisk applications can be called as
app.application_name.
Regards,
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin
Sent: Friday, February 18, 2011 4:33 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] lua -asterisk manual
Please could someone advise good manual for using lua for asterisk dialplan.
There is not much docu about it.
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Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2
Are you on CentOS? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin Sent: Friday, February 18, 2011 7:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2 On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif fai...@vopium.com wrote: Did you checked if you extension.ael doesn't have syntax error? I think there is no error. I loaded the standard ael first (provided by asterisk) then my test config, got the same result. Did you upgraded anything after last compile? No. I just took ver 1.6.2.16.1 , compiled with ael support got this error. then decided to check with ver 1.8.2. Error remained the same. Or Try a clean recompile Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin Sent: Friday, February 18, 2011 4:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2 Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47] WARNING[7412]: loader.c:839 load_resource: Module 'pbx_ael.so' could not be loaded. I did not find in google what it could be and what should be done to solve this. I also tried the same on ast ver 1.8.2.3, got the same. I am usind debian as OS and install asterisk from sources that I took on digium site. Did anyone have the same issue? Regards, Kate -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Fwd: cmd MySQL
If you are using asterisk 1.8.x you don't need to type \ for spaces you can write simple query and use spaces as normal it will work fine. Faisal From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe Figueiredo Sent: Friday, February 18, 2011 11:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fwd: cmd MySQL -- Forwarded message -- From: Felipe Figueiredo felipe.figueired...@gmail.com Date: Fri, Feb 18, 2011 at 4:03 PM Subject: Re: [asterisk-users] cmd MySQL To: Gerald A geraldabli...@gmail.com - Executing [200@teste:2] MYSQL(Console/dsp, Query resultid 1 SELECT\ ramal\ FROM\ colaboradores\ WHERE\ ramal=200) in new stack [Feb 18 16:01:42] WARNING[7749]: app_mysql.c:393 aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near '\ ramal\ FROM\ colaboradores\ WHERE\ ramal=200' at line 1 hi Gerald, look, the error is the same. Eveng changing the / for \ ... On Fri, Feb 18, 2011 at 4:00 PM, Gerald A geraldabli...@gmail.com wrote: Hi Felipe, On Fri, Feb 18, 2011 at 12:56 PM, Felipe Figueiredo felipe.figueired...@gmail.com wrote: -- Executing [200@teste:2] MYSQL(Console/dsp, Query resultid 1 SELECT/ ramal/ FROM/ colaboradores/ WHERE/ ramal=200) in new stack [Feb 18 15:55:13] WARNING[7696]: app_mysql.c:393 aMYSQL_query: aMYSQL_query: mysql_query failed. Error: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near '/ ramal/ FROM/ colaboradores/ WHERE/ ramal=200' at line 1 I'm not Asterisk-MySQL guru, but shouldn't the / be \? I'm guessing you are trying to keep a string together here, but maybe I'm mistaken. Thanks, Gerald -- _ -- 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
Re: [asterisk-users] How to know Caller's last position in Queue?
If you use Asterisk 1.8.x you can have this in channel vars and can collect
and add to DB or file on h extension.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Man
Sent: Wednesday, February 16, 2011 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to know Caller's last position in Queue?
Hi group,
I have a simple call center scenario set up on Asterisk. Customer calls the
DID and gets placed in Queue waiting for their turn to talk to the available
agent.
Sometimes Customer hangs up in between and in this case I want to get the
last position of customer in Queue.
I know there is a variable called ${QEORIGINALPOS} that gives us original
position of caller in Queue, but there doesn't seem to have something
similar for exit position.
Am I missing something?
Thanks,
--AsteriskMan
--
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Re: [asterisk-users] how to diable echo cancellation for sip?
It is in client but not in asterisk sip channel From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 4:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to diable echo cancellation for sip? Hello, can anyboby tell me, how can I disable the echo cancellation for sip? thx a lot... best regards, Felix -- _ -- 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
Re: [asterisk-users] function Echo() doesn't work
Did you executed Answer() before it? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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
Re: [asterisk-users] function Echo() doesn't work
Check if you have incoming SIP call in supported codec or send CLI log for further troubleshooting. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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 -- _ -- 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
Re: [asterisk-users] function Echo() doesn't work
I faced same issue for sipgate but got it resolved by allowing all codec in sipgate peer config. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work == Using SIP RTP CoS mark 5 -- Executing [1174614@von-voip-provider:1] Answer(SIP/sipgate-account-, ) in new stack -- Executing [1174614@von-voip-provider:2] Echo(SIP/sipgate-account-, ) in new stack == Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on 'SIP/sipgate-account-' here is the log. It is as same as I got from CAPI and Datacard. I just didn't hear the echo from SIP connection. 2011/2/16 Faisal Hanif fai...@vopium.com Check if you have incoming SIP call in supported codec or send CLI log for further troubleshooting. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] How to know Caller's last position in Queue?
You have enable following in queue configuration,
setinterfacevar=yes
setqueueentryvar=yes
setqueuevar=yes
and you will find your data in following variables,
${QEORIGINALPOS} will have position when caller enter the queue.
${QUEUEPOSITION} will have position when caller left the queue.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Man
Sent: Wednesday, February 16, 2011 5:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know Caller's last position in Queue?
Hi Hanif,
I indeed use 1.8 .0 but couldn't find the channel variable for caller's
last position in queue anywhere in documentation.
Would you please let me know the channel variable name?
Thanking you.
On Wed, Feb 16, 2011 at 4:40 PM, Faisal Hanif fai...@vopium.com wrote:
If you use Asterisk 1.8.x you can have this in channel vars and can collect
and add to DB or file on h extension.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Man
Sent: Wednesday, February 16, 2011 3:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to know Caller's last position in Queue?
Hi group,
I have a simple call center scenario set up on Asterisk. Customer calls the
DID and gets placed in Queue waiting for their turn to talk to the available
agent.
Sometimes Customer hangs up in between and in this case I want to get the
last position of customer in Queue.
I know there is a variable called ${QEORIGINALPOS} that gives us original
position of caller in Queue, but there doesn't seem to have something
similar for exit position.
Am I missing something?
Thanks,
--AsteriskMan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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_
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Re: [asterisk-users] function Echo() doesn't work
Did you make any peer for sipgate if yes then do for that peers. Please also note that disallow line should be before allow line. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 6:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work I tried to set allow=all in sip.conf. But it still doesn't work. 2011/2/16 Faisal Hanif fai...@vopium.com I faced same issue for sipgate but got it resolved by allowing all codec in sipgate peer config. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work == Using SIP RTP CoS mark 5 -- Executing [1174614@von-voip-provider:1] Answer(SIP/sipgate-account-, ) in new stack -- Executing [1174614@von-voip-provider:2] Echo(SIP/sipgate-account-, ) in new stack == Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on 'SIP/sipgate-account-' here is the log. It is as same as I got from CAPI and Datacard. I just didn't hear the echo from SIP connection. 2011/2/16 Faisal Hanif fai...@vopium.com Check if you have incoming SIP call in supported codec or send CLI log for further troubleshooting. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] function Echo() doesn't work Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS Stick). Just only no echo on SIP. Any suggestion? 2011/2/16 Faisal Hanif fai...@vopium.com Did you executed Answer() before it? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong Sent: Wednesday, February 16, 2011 4:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] function Echo() doesn't work Hi guys, the function Echo() did work on CAPI, but doesn't work for SIP connection. Can anybody help? thanks a lot. best regards, Felix -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Play one audio file to the called part before the Dial() command
You can do it using callback files or AMI. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Songtao Yu Sent: Wednesday, February 16, 2011 6:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Play one audio file to the called part before the Dial() command Hi, I am not sure if it is doable: 1. We originate one call from Asterisk 2. Asterisk plays one audio file to the called part when the called part picks up the phone. 3. Asterisk establish one real connection between the caller part and the called part. Thanks, Songtao Yu -- _ -- 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
Re: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP
Well a quick n easy fix for you is you can configure you call sending peers to use username secret in INVITE. As far as I know it possible in almost all CISCO, Avaya and all other standard Gateway and SBCs which follows full SIP RFCs. If you can't do it then you need to use curl as realtime engine instead of MySQL. It will call a URL for each SIP request which you can handle with flexibility in your CGI scripts with apache. But be careful as per my experience asterisk 1.6 with curl as realtime engine can handle a max of 120 registration in parallel if registration refresh time is 120 seconds. Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo Carvalho Sent: Wednesday, February 16, 2011 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- 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
Re: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP
I have played a lot on this issue with asterisk config but in realtime it doesn't supported static peers with version 1.6.2.14. From: Ricardo Carvalho [mailto:rjcarvalho.li...@gmail.com] Sent: Wednesday, February 16, 2011 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Faisal Hanif Subject: Re: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP Isn't this a limitation that can be surpassed with some configuration that I'm lacking in my sip.conf or extensions.conf of my asterisk? Ricardo. On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif fai...@vopium.com wrote: Well a quick n easy fix for you is you can configure you call sending peers to use username secret in INVITE. As far as I know it possible in almost all CISCO, Avaya and all other standard Gateway and SBCs which follows full SIP RFCs. If you can't do it then you need to use curl as realtime engine instead of MySQL. It will call a URL for each SIP request which you can handle with flexibility in your CGI scripts with apache. But be careful as per my experience asterisk 1.6 with curl as realtime engine can handle a max of 120 registration in parallel if registration refresh time is 120 seconds. Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo Carvalho Sent: Wednesday, February 16, 2011 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with 407 Proxy Authentication Required, I assume because they don't carry the registered contact registration!!! My SIP contacts have type=friend and all inbound calls not coming from my registered phones fall in the default context without authentication, so that someone in the Internet be able to call freely through the Internet anyone in my server's dial plan. Some ideas? Regards, Ricardo Carvalho. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] pipe audio stream to external application
EAGI could be your target application. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Wednesday, February 16, 2011 11:01 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] pipe audio stream to external application Hi, I'd like to know if there's an easy way of doing the following: SIP phone dials a custom feature code in Asterisk, call gets answered within a custom context (Answer()), anything that the caller says should be redirected/piped to an external application. Something like monitor except audio should be sent live. More like app_ices (or app_ezstream if that existed) but for a generic app. Thanks Vieri -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Asterisk on a USB with persistence
You can simply use Portable LinuxLive USB Creator 2.6 or grub4dos. And make your USB bootable by any Linux Live ISO. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of logan Sent: Wednesday, February 16, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk on a USB with persistence Hi, I'm looking to get an ISO of FreePBX or AsteriskNOW installed on a USB that I can boot from and also be able to save my changes. Is this possible? My search on web doesn't seem to find anything useful. For now I don't have the option of having a spare machine or creating a partition on my existing one for my experiments with Asterisk. My end goal is to have chan_mobile configured and see if I can make calls through my cellphone using that. Thanks, Hitesh -- _ -- 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
Re: [asterisk-users] trunks and phones registered from the same IP
In case of asterisk you simply can't accept registration from an IP which you have mentioned as static host for IP authentication. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali Sent: Tuesday, February 15, 2011 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] trunks and phones registered from the same IP really it's too difficult to understand, please explain more clear On Tue, Feb 15, 2011 at 5:17 AM, Ricardo Carvalho rjcarvalho.li...@gmail.com wrote: Hi, How can I configure my asterisk server so that I can receive incomming calls comming from the same IP from where my server also receives phone registrations? The problem is that since the moment any extension registers at that IP (actually I have a registration proxy running at that IP), asterisk no more accepts calls coming from a SIP trunk I also have at that IP, replying back with 401 Unauthorized. Any ideas? Thanks, Ricardo. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] unregistered trunks and registered phones coming from the same IP
You need to use relay request in your SBC instead of forward. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali Sent: Tuesday, February 15, 2011 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] unregistered trunks and registered phones coming from the same IP please send your sip.conf, is any NAT procedure implemented in your network? On Mon, Feb 14, 2011 at 10:16 PM, Ricardo Carvalho rjcarvalho.li...@gmail.com wrote: Hi, I manage an SBC which stands between my company server farm and some SIP telco trunks. The system works fine, for inbound and outbound calls. Now I've configured the SBC to also act as a registration proxy, forwarding SIP registrations coming from the Internet to my asterisk servers. It all seems fine, but it doesn't work well, because by the time at least one phone registers through the SBC to some asterisk server (lets say, server_A), future incoming calls coming from my SIP telco trunks to my server_A got refused by the asterisk running on that server, with 401 Unauthorized messages back to the SBC. Seems like that since the moment asterisk binds some contact to the IP of the SBC, because it registered through it, from that moment, asterisk only accepts calls from that IP if those INVITEs carry correct registration to my server (even if those calls came from my SBC, a trusted trunk, not registered in asterisk). My phones are configured with type=friend. I've also tried type=peer and type=user, but it doesn't solve the problem. Any ideas to fix this? Best regards, Ricardo. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] DTMF not detected, time out
Check if dtmfmode is properly set on SIP trunk ask with your carrier which dmtfmode they support. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, February 16, 2011 5:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DTMF not detected, time out Hi, I encounter this problem recently after quite some months of my asterisk. I have a SIP trunk for dial in and out. When dial-in, it matches the callerid number and decides. If matched, it will either go into an a very brief IVR. The IVR allows caller to dial internal extension. All along it is working well both from outside call and internal users. Now for unknown reason, it fails with a timeout and hangup. It is the only message I can see at the console. But internal user can do this without any problem. Appreciate if someone can help. CK -- _ -- 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
Re: [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3
You may need to share your LUA code and the extension your call is need to execute. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlo Pires Sent: Wednesday, February 16, 2011 3:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3 But when I try to call one extension created with lua I got a message telling that extension doesnt exist on default context. Am I missing something? 2011/2/15 Tilghman Lesher tilgh...@meg.abyt.es: On Tuesday 15 February 2011 11:06:32 Carlo Pires wrote: Hi, After compiling a installing asterisk 1.8.2.3 I wanted to play with lua but I noticed that extensions created in extensions.lua was not being registered with asterisk. uga1*CLI dialplan show [ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ] 's' = 1. NoOp() [app_queue] [ Context 'parkedcalls' created by 'features' ] '700' = 1. Park() [features] [ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ] 's' = 1. NoOp() [app_dial] [ Context 'local' created by 'pbx_lua' ] Alt. Switch = 'Lua/' [pbx_lua] [ Context 'demo' created by 'pbx_lua' ] Alt. Switch = 'Lua/' [pbx_lua] [ Context 'default' created by 'pbx_lua' ] Alt. Switch = 'Lua/' [pbx_lua] -= 3 extensions (3 priorities) in 6 contexts. =- uga1*CLI uga1*CLI dialplan show demo [ Context 'demo' created by 'pbx_lua' ] Alt. Switch = 'Lua/' [pbx_lua] -= 0 extensions (0 priorities) in 1 context. =- uga1*CLI Need I enable something to get lua extensions to be created? No, that's how Lua extensions work, with the switch statement. Your extensions are still being evaluated by Lua. The only difference is that pbx_lua now doesn't see any need to create extensions, because it will see every extension when it hits the switch. -- Tilghman -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] DTMF not detected, time out
Ask with you SIP carrier which dtmfmode they are using on their end and use same on asterisk side. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, February 16, 2011 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF not detected, time out In the past it was set as auto and worked. I change to RFC2833 but did not work. How can I check further? On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif fai...@vopium.com wrote: Check if dtmfmode is properly set on SIP trunk ask with your carrier which dmtfmode they support. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, February 16, 2011 5:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DTMF not detected, time out Hi, I encounter this problem recently after quite some months of my asterisk. I have a SIP trunk for dial in and out. When dial-in, it matches the callerid number and decides. If matched, it will either go into an a very brief IVR. The IVR allows caller to dial internal extension. All along it is working well both from outside call and internal users. Now for unknown reason, it fails with a timeout and hangup. It is the only message I can see at the console. But internal user can do this without any problem. Appreciate if someone can help. CK -- _ -- 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 -- _ -- 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
Re: [asterisk-users] DTMF not detected, time out
You can also append add dtmf logging to cosole and see if dtmf is coming from carrier. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, February 16, 2011 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF not detected, time out In the past it was set as auto and worked. I change to RFC2833 but did not work. How can I check further? On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif fai...@vopium.com wrote: Check if dtmfmode is properly set on SIP trunk ask with your carrier which dmtfmode they support. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, February 16, 2011 5:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DTMF not detected, time out Hi, I encounter this problem recently after quite some months of my asterisk. I have a SIP trunk for dial in and out. When dial-in, it matches the callerid number and decides. If matched, it will either go into an a very brief IVR. The IVR allows caller to dial internal extension. All along it is working well both from outside call and internal users. Now for unknown reason, it fails with a timeout and hangup. It is the only message I can see at the console. But internal user can do this without any problem. Appreciate if someone can help. CK -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Connect Asterisk to a cell phone
Hi, Your question is not clear but below are possible answers to your question, If you want to attach you cell-phone to asterisk you can simply use chan_mobile. Using Bluetooth with chan_mobile you can connect your Cell-Phone as FXO and your handsfree as FXS port to asterisk. If you are asking about a GSM to SIP gateway then yes there are number of product available that can hold 1-256 SIM and register as SIP gateway to asterisk for incoming and outgoing calls. If you are asking about GSM PCI card then also yes there are PCI cards available for GSM/CDMA/HSPDA for 1-16 SIMs. Can pluged to asterisk PBX machine and used as FXO device. Regards, Faisal Hanif From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of logan Sent: Wednesday, February 16, 2011 10:49 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Connect Asterisk to a cell phone Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. -- _ -- 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
Re: [asterisk-users] Ported Asterisk in Android
Well I think you need major changes as application in android run in sandbox instead of direct Linux APIs. Till now no news on it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil Sent: Monday, February 14, 2011 6:46 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ported Asterisk in Android waiting for replys.. On 02/11/2011 02:20 PM, Nikhil wrote: Thanks for reply. Any other suggestions . On 12/20/2010 05:52 PM, Service clients - VDI CENTER wrote: i believe there is a way to do it using asterisk and flashphoner ++ 2010/12/20 Gilles codecompl...@free.fr On Fri, 17 Dec 2010 15:51:33 +0530, Nikhil d.nik...@cem-solutions.net wrote: Does anyone ported Asterisk to Android OS .please give details www.servalproject.org http://www.servalproject.org/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 -- Cordialement Gabriel 09 79 94 71 13 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error
Better to report a BUG to cisco. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller Sent: Monday, February 14, 2011 6:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error Sensitivity: Confidential Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file. I have not been able to locate anything in google about this kind of issue and am at a loss as to what in the world is the issue. I have asterisk 1.8.2.2 installed with the FreePBX module with a 7960 just recently flashed to 8.12. Not sure what else you all may need but any help would be greatly appreciated. Respectfully, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. image001.gif-- _ -- 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
Re: [asterisk-users] issue with some numbers
You may need to provide some more scenario detail From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Monday, February 14, 2011 7:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] issue with some numbers Hello all I have a small issue with some mobiles numbers when I call these numbers using asterisk I have all the time answer machine. But when I call these numbers using my mobile or another phone there is no problem. Any help will be appreciated -- _ -- 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
Re: [asterisk-users] Early audio SIP sequence order question
Go here http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial and use proper parameters to dial command to pass early media. -Original Message- From: Benoit Panizzon Sent: Thursday, February 10, 2011 4:08 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Early audio SIP sequence order question Hello We have quite some problems with early audio with our asterisk 1.6.2.15 What we observe is: Asterisk - Carrier PBX Asterisk:Invite(+sdp) = Carrier Carrier starts to send RTP Audio (ignored by Asterisk) Asterisk = Carrier:100 Trying Asterisk = Carrier:180 Ringing Asterisk signals Ringing to the caller which in turn generated the ringing tone (still ignoring the early audio sent by the carrier). Asterisk = Carrier:200 OK(+sdp) Asterisk:ACK = Carrier Asterisk starts to send RTP Audio to Carrier Only now Asterisk starts playing Audio to the caller. This causes quite troubles, as the price of a value added number is announced in early audio in switzerland, giving the caller a chance to hang up before the call is established. But the caller connected to asterisk does not hear that early audio announcement. Is this an asterisk bug, or should the carrier have signaled 183 Session Progress instead of 180 Ringing? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz Web http://www.imp.ch __ -- _ -- 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 -- _ -- 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
Re: [asterisk-users] CDR with unix time.
Well. I suggest to use DB function instead of modifying asterisk source. You can add one additional column and write and after-insert trigger in your cdrs table which convert dattime to your required format and update the value of added column. From: Rodrigo Lang Sent: Thursday, February 10, 2011 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDR with unix time. Good morning everyone. I wonder if it is possible, without touching the source code, to Asterisk save the cdr with date in unix time instead of the default date. It's possible? Thanks in advance, -- Rodrigo Lang Opening your mind - Just another Open Source site -- _ -- 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-- _ -- 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
Re: [asterisk-users] zaptel/dahdi settings for singtel E1 line
The settings you are asking varies in different countries and providers. You need to contact you provider for it. From: Roi Stork Sent: Thursday, February 10, 2011 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] zaptel/dahdi settings for singtel E1 line Anyone here who has configured zaptel/dahdi for a singtel E1 line? What are the settings for coding, framing, line type and switchtype? -- _ -- 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-- _ -- 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
Re: [asterisk-users] dial option 'g' not working
There are some flags in general settings of dialplan which enable/disable modify this behaviors of dialplan. Have a look on sample extensions.conf for general tab settings. I will see if I can have time today to tell you exact parameter name. From: Dovid Bender Sent: Thursday, February 10, 2011 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dial option 'g' not working Hi, I had the same issue as well but for some reason I was unable to reproduce. Please have a loo at: https://issues.asterisk.org/view.php?id=18682 Regards, Dovid - Original Message - From: M S To: asterisk-users@lists.digium.com Sent: Wednesday, February 09, 2011 06:11 Subject: [asterisk-users] dial option 'g' not working Hi, I'm trying to get my dialplan to continue executing in the current context after a third-party is called and hangs up. It seems like it should be straightforward but it's not working. Here's what I have in extensions.conf: exten = 333,1,Answer() exten = 333,n,Playback(hello) exten = 333,n,Dial(SIP/1999222@sipcarrier,,g) exten = 333,n,Playback(hello) exten = 333,n,Playback(hello) exten = 333,n,Playback(hello) exten = 333,n,Hangup() The 999222 number is dialed, but after that party hangs up, there's just dead air. No hello's are played and nothing seems to be happening. What am I doing wrong? Thanks, MS -- -- _ -- 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 -- _ -- 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-- _ -- 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
Re: [asterisk-users] Base memory usage
Hi, 1-TuneUp your setting in /etc/dafult/asterisk 2-Stop l;oadng all not required modules by adding noload = modulename.so lines to /etc/asterisk/modules.conf Regards, Faisal On 12/31/2010 7:59 AM, Larry Wimble wrote: Asterisk gurus I just installed asterisk 1.8.1.1 along with FreePBX on a fairly small VPS (512mb standard, 512mb burst). I note that the asterisk process is using about 209mb of memory just doing nothing (not configured to do anything yet) In contrast to this, my 1.6.1.2 installation from a little over a year ago uses only 40mb and it's fully configured and running with about 4 months of uptime (2 trunks, 4 channels, 3 DIDs, and 4 extensions.) Any ideas on how I can get the memory consumption down on my new installation, or is it time to downgrade to the older version? Thanks, Larry -- _ -- 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 -- _ -- 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
Re: [asterisk-users] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone
Hi, I have used 4-PRI card from atcom.cn and it works perfectly for me. Regards, Faisal +923214059996 On 12/27/2010 12:25 PM, Asim Amin wrote: Hello All, Anyone who has experience using Digium analog card clones from any of the following: 1. Zycoo 2. CTVON 3. Chinaroby 4. Etross 5. Immediate IT (IIT) 6. Realtone and can give review which one is good quality with easy configuration and error free running. Also since some of these manufacture only analog cards, does anyone have any experience using these in a single system with digital cards from other manufacturers like Openvox? -- Asim Amin Partner Technical Manager, Telco Division Horizon Technologies Cell: +92-323-3314151 E-mail: a...@horizontech.biz mailto:a...@horizontech.biz Web: http://horizontech.biz http://horizontech.biz/ http://hostht.com http://hostht.com/ -- _ -- 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 -- _ -- 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
Re: [asterisk-users] differential billing
Hi Abdul-Basit, If you need only different intervals of billing you can easily do it using any AGI as we are doing it in Perl AGIs using post call billing. But if you need realtime billing then the most stable and flexible option is to use FastAGI+ AMI. I have tested it in JAVA and it worked for me up to a load 100 calls. It may work more but I haven't tested it. Asterisk and Billing-Server was running on separate machines. For further help you can call me (as you know my number :P). Regards, Faisal Hanif -- _ -- 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
Re: [asterisk-users] CDR display in minute
use CACTI On 9/23/2010 3:10 PM, Mickael MONSIEUR wrote: Hello, I want to graphically display the number of calls per minute to an extension. The programs I have found it possible to do so but the average is done on time or day ... I use Mysql CDR Thank you, Mickael -- _ -- 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
Re: [asterisk-users] A way to check against a list of numbers?
Hi, An intelligent way is to maintain numbers list in any Database (could be SQlite if you don't want to use proper DB engine) then use ODBC-Function if the number is there and decide routing. 2nd option is to use Perl AGI with DBI::CSV and manage numbers list in a CSV file. Regards, Faisal Hanif On 9/11/2010 4:47 PM, Olivier wrote: 2010/9/10 Hose hose+aster...@bluemaggottowel.com mailto:hose%2baster...@bluemaggottowel.com Does anyone have a suggestion on how to handle this? For example, if I have a list of numbers that I want to go out a certain sip channel and another that I want to go out the dahdi device, is there a way to do this? None of the numbers will fit into a pattern, so just plain pattern matching won't do. The most straightforward way would be to just define explicit patterns. Obviously that works, but doesn't seem scalable in terms of maintenance. Ideally there should be a variable or list of numbers, How many numbers do you plan ? and the dialplan logic jumps into a subroutine that checks if the dialed number is on the list, then routes accordingly. Does anyone have any suggestions as to how to approach that, or if they have a entirely different way in mind? hose -- _ -- 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 -- _ -- 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
Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Allow anonymous SIP and enable debug then check if calls coming from same IP which you have configured in peer? Regards, Faisal Hanif// On 9/11/2010 8:07 AM, bruce bruce wrote: Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing sip set debug peer PROVIDER: Sending to 123.123.123.123 : 5060 (no NAT) That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see the ACK, REQUEST and ACCEPT of sip debug just fine. This is Elastix with Asterisk 1.4.33.1 Any thoughts? Thanks -- _ -- 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
Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon
did you copied rc.redhat.asterisk script from contrib/init.d/ forlder to /etc/init.d/ folder? Regards, Faisal Hanif On 8/16/2010 2:28 PM, unsero...@aol.com wrote: No ideas? Sorry but I'm new to Linux and I am wondering why I can't stop or start the deamon the way it works with 1.6.1.20. I installed Asterisk 1.8 with all defaults set. Maybe something is missing in any conf file? Make sure it starts without the daemon. Try asterisk -cvvv. Does it start then? sean -- Yes, without the daemon it starts and i don't see any errors. It also starts automatically after a system boot. But I am wondering why I can't stop|start|restart using /etc/init.d/asterisk start|stop|restart like in 1.6? -- _ -- 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
Re: [asterisk-users] How to determine which party hangup th e call? cause of Hang-up needed.
Hi,
Your dial-plan could be like this,
here you will dial
EXTEN = _X,1,NoOp
EXTEN = _X,n,Set(WHOHAVEHANGED=CALLER)
EXTEN = _X,n,Dial(ZAP/xyz)
if caller hanged below line will never be executed because control will
go to h extension.
EXTEN = _X,n,Set(WHOHAVEHANGED=CALEE)
EXTEN = h,1,NoOp(${WHOHAVEHANGED} have hanged the call reason is
${HANGUPCAUSE})
Regards,
Faisal Hanif
On 8/12/2010 12:29 AM, bruce bruce wrote:
Sorry, I am not following:
*//**/read the value of var ${HANGUPCAUSE} next line to dial command./*
*/
/*
*/Where is that value? Next to dial you mean right when the call was
placed? or check next few lines to find HANGUP cause?/*
*/
/*
*/Note: This is using ZAP (analogue) and not PRI./*
*/
/*
*/Thanks,/*
*/Bruce
/*
On Wed, Aug 11, 2010 at 12:33 AM, Faisal Hanif fai...@vopium.com
mailto:fai...@vopium.com wrote:
read the value of var ${HANGUPCAUSE} next line to dial command.
Regards,
Faisal Hanif
/VoIP Manager
/**Vopium A/S
On 8/10/2010 9:51 PM, bruce bruce wrote:
Hi Everyone
Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines
to Bell Canada.
User claims that call hangup without any interferance to the
phone set.
Is there ANYWAY to find out which party hang-up the call or if
the call was cut-off due to other reasons?
I checked the *asteriskcdrb* table and it's pretty much useless
in this case as it only logs the duration and other properties
but not cause of the Hangup.
/var/log/asterisk/full
[Jul 10 10:37:02] VERBOSE[29366] logger.c: == Manager 'admin'
logged off from 127.0.0.1
[Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Executing
[...@macro-dialout-trunk:1] Macro(SIP/1007-069a,
hangupcall|) in new stack
[Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Executing
[...@macro-hangupcall:1] GotoIf(SIP/1007-069a, 1?skiprg) in
new stack
[Jul 10 10:37:09] VERBOSE[29348] logger.c: -- Goto
(macro-hangupcall,s,4)
Thanks,
Bruce
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