Re: [Asterisk-Users] H.264 and Asterik?
Kevin: Thanks for the info, I think I will buy the video phones Erick W. - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 19, 2006 6:18 PM Subject: Re: [Asterisk-Users] H.264 and Asterik? Erick Weber V. wrote: Dose someone know if the latest version of asterisk support H.264? Asterisk SVN trunk (which will become Asterisk 1.4) supports H.264, and I have a Grandstream H.264 phone on my desk right now which I am testing with it (and it works fine!). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.264 and Asterik?
Hello: Dose someone know if the latest version of asterisk support H.264? I´ll like to buy some Grandstream video phones but it uses H.264 Thanks Erick Weber ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.264
Matt: Do you know if it can transcode between H.263 and H.264 Thanks for the info. Erick W - Original Message - From: Matt Riddell [NZ] [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 11, 2006 12:55 AM Subject: Re: [Asterisk-Users] H.264 Erick Weber V. wrote: Hello: Does someone know if asterisk supports H.264 video codec Find attached cvs commit note, so yes, if you get the latest trunkversion of asterisk you will get h264. Subject: [svn-commits] trunk - r7855 in /trunk: ./ channels/ formats/ include/asterisk/ From: svn-commits@lists.digium.com Date: Sat, 07 Jan 2006 17:54:23 - To: [EMAIL PROTECTED], svn-commits@lists.digium.com To: [EMAIL PROTECTED], svn-commits@lists.digium.com Author: markster Date: Sat Jan 7 11:54:22 2006 New Revision: 7855 URL: http://svn.digium.com/view/asterisk?rev=7855view=rev Log: Add support for H.264 with SIP and recording -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.264
Hello: Does someone know if asterisk supports H.264 video codec Thanks Erick W. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 videoconferencing with asterisk?
Hello: I´ll like to know if asterisk is capable of making H.323 videoconferencing and if it can also transcode fromH.323 to SIP Any help will be appreciate Tanks Erick W. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?
For me to - Original Message - From: Mat Stace, Colewood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 15, 2005 5:46 PM Subject: Re: [Asterisk-Users] Voipbuster blocking Asterisk/IAX connections? As of 22:45 GMT it's working for me Jerry Glomph Black wrote: This service has been working well lately, but as of this morning is promptly blowing off IAX connections with the dreaded 'No Authority Found' error. Any concrete info greatly appreciated! Dr G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.10.7/70 - Release Date: 11/08/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA186 can not generate dtmf
Hello: I have problems sending dtmf signal to an ATA186 my configuration is: ATA186 -- asterisk -- ATA186 -- FXS to FXO Converter -- PSTN The ATA186 are set to send dtmf RFC2833, but it seems that the ATA186 can't generate dtmf so I can dial to a PSTN number. Is there a setting that can fix my problem, inband dtmf does not work because I'm using G729 codec Thanks Erick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Send dtmf to ata186!!!
Hello: I have problems sending dtmf signal to an ATA186 my configuration is: ATA186 -- asterisk -- ATA186 -- FXS to FXO Converter -- PSTN The ATA186 are set to send dtmf RFC2833, but it seems that the ATA186 can't generate dtmf. Is there a setting that can fix my problem, inband dtmf does not work because I'm using G729 codec Thanks Erick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] aah and astcc
Darren: Thanks for your interest I would like that once you have been verified you can use aah dial plan so you can get all the reports for the astcc calls Thanks for your help Erick Weber - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 06, 2005 8:26 PM Subject: Re: [Asterisk-Users] aah and astcc How exactly are you thinking. So that a certain aah extension points to it or so that once you have been verified you can call aah extensions? Darren Erick Weber V. wrote: Hello: Does anyone know how to incorporate astcc to aah so it will use amah extensions. Any help will be appreciate Thanks Erick W. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] aah and astcc
Hello: Does anyone know how to incorporate astcc to aah so it will use amah extensions. Any help will be appreciate Thanks Erick W. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729
Title: Untitled Document Hi, The Sipura SPA-2000 can only support one G729 call Regards Erick - Original Message - From: David To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Friday, June 17, 2005 11:33 AM Subject: [Asterisk-Users] G729 Hi All, I have configured Line1 (2011)and Line2(2012)in SipuraSPA-2000 (latest Firmware)to use G729. In sip.conf I have set disallow=all, allow=g729 IfLine1 is in use by an agent, then Line2 won't work and viceversa (Inbound Calls Only).I have 40 license for G729. so there shouldn't be any issue with the license. I'm getting the following error msg: -- Called 2012 -- Got SIP response 488 "Not Acceptable Here" back from 192.168.10.103 == No one is available to answer at this time (1:0/0/0) == Auto fallthrough, channel 'IAX2/[EMAIL PROTECTED]:4569-5' status is 'NOANSWER' -- Hungup 'IAX2/[EMAIL PROTECTED]:4569-5' If I change 2012 to ULAW, it works fine. It seems that I can't have two lines configured as a G729. Do you guys have any idea why this happening? Regards, ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] play gsm files in windows
Use Apple QuickTime Best Regards Erick W. - Original Message - From: Brett, Gary [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 23, 2005 9:34 AM Subject: [Asterisk-Users] play gsm files in windows Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. Any help would be greatly appreciated cheers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] play gsm files in windows
See the Wiki: http://www.voip-info.org/wiki-Asterisk+sound+files you can doit with SOX on yor server or Linux box This is how you convert wav files to gsm files used by Asterisk $ sox inputfile.wav -r 8000 -c 1 outputfile.gsm resample -ql Best Regads Erick W - Original Message - From: Brian C. Fertig [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 23, 2005 10:19 AM Subject: RE: [Asterisk-Users] play gsm files in windows Eric, Do you know of one that can convert or record? .o---o. Brian Fertig NOC/Network Engineer Systems Engineer Planet Telecom, Inc. Tampa, FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Monday, 23 May, 2005 11:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] play gsm files in windows Use Apple QuickTime Best Regards Erick W. - Original Message - From: Brett, Gary [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 23, 2005 9:34 AM Subject: [Asterisk-Users] play gsm files in windows Does anybody know of a WINDOWS application (preferably freeware) that will simply playback asterisk GSM sound files, I don't want to record them, just playback the ones that are currently there. Any help would be greatly appreciated cheers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email was scanned by: Mcafee GroupShield ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help Please!!!!
Thanks, I will begin my testing Erick - Original Message - From: Race Vanderdecken [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, February 16, 2005 8:18 PM Subject: RE: [Asterisk-Users] Help Please Greetings Mr. Weber, Remember the rule in mathematics that is much easier to solve for one variable. You stateed you are having a problem with the 1088 extension. If look like you are trying to make a call from the 404 extension to the 1088 extension. 1. If you have 6 ATA's running shut 5 of them off. Test each one separately. Then turn one on at a time and see the problem can be traced to one ATA 2. You are getting sent an authorization request from asterisk to the 1088 extension. WWW-Authenticate: Digest realm=asterisk, nonce=0711b1d6 Make sure you don't have any of the secret= or the md5secret= stuff set in the sip.conf, until you can get each phone to talk in the open. Then change, one, 1, uno, phone at a time. 3. If you have a SIP phone that is not an ATA then set it up and try to dial the 1088 and see if you get the same thing. 4. Do a sip show users to make sure the 1088 is registered with asterisk. 5. Do the normal, things don't work dance, by unplugging the phone and reconnecting a different phone to the ata. Change the power suplly with another ata. Change the RJ45 patch cable. Try a different port in the switch or wall. Swap one of the known working ATA and change it to the 1088 ata. 6. Go to lunch and have a beer. Find a new job and settle down with a good woman. Leave telecom and go into organic farming. Race The Tyrant Vanderdecken [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Wednesday, February 16, 2005 2:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Help Please Importance: High I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is that one of them is dropping calls an I can't figure out what is the problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem. Any help will be appreciate Thanks Erick Weber VoIP*CLI sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer RTP is at port 192.168.1.69:0 -- Executing Dial(SIP/404-cbc9, SIP/1088|60|tr) in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX s=session c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 17506 RTP/AVP 18 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - (NAT) to 201.133.170.82:5060 -- Called 1088 -- SIP/1088-ec82 is ringing Found RTP audio format 18 Found RTP audio format 101 Peer RTP is at port 192.168.1.2:0 Found description format G729 Found description format telephone-event Capabilities: us - 0x100(G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x100(G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.1.2, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED];tag=939809556 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 201.133.170.82:5060 -- SIP/1088-ec82 answered SIP/404-cbc9 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 Using latest request as basis request Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1
[Asterisk-Users] Help Please!!!!
I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is that one of them is dropping calls an I can't figure out what is the problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem. Any help will be appreciate Thanks Erick Weber VoIP*CLI sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer RTP is at port 192.168.1.69:0 -- Executing Dial(SIP/404-cbc9, SIP/1088|60|tr) in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 164 v=0 o=root 1679 1679 IN IP4 XXX.XXX.XXX.XXX s=session c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 17506 RTP/AVP 18 a=rtpmap:18 G729/8000 a=silenceSupp:off - - - - (NAT) to 201.133.170.82:5060 -- Called 1088 -- SIP/1088-ec82 is ringing Found RTP audio format 18 Found RTP audio format 101 Peer RTP is at port 192.168.1.2:0 Found description format G729 Found description format telephone-event Capabilities: us - 0x100(G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x100(G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.1.2, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK642900c4;rport From: Weber Automundo sip:[EMAIL PROTECTED];tag=as4da46cda To: sip:[EMAIL PROTECTED];tag=939809556 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 201.133.170.82:5060 -- SIP/1088-ec82 answered SIP/404-cbc9 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 -- Attempting native bridge of SIP/404-cbc9 and SIP/1088-ec82 Using latest request as basis request Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=0711b1d6 Content-Length: 0 to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Using latest request as basis request Sending to 192.168.1.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 201.133.170.82:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.2:5060;received=201.133.170.82;rport=5060 From: sip:[EMAIL PROTECTED];user=phone;tag=3858230914 To: sip:[EMAIL PROTECTED];user=phone;tag=as601a996c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 120 Contact: sip:[EMAIL PROTECTED];expires=120 Date: Wed, 16 Feb 2005 00:43:46 GMT Content-Length: 0 to 201.133.170.82:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:201.133.170.82 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK0972cae7 From: asterisk sip:[EMAIL PROTECTED];tag=as59adf4c2 To: sip:201.133.170.82 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 201.133.170.82:5060 Destroying call '[EMAIL PROTECTED]' set_destination: Parsing sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp for address/port to send to set_destination: set destination to 192.168.1.2, port 5060
[Asterisk-Users] Maximum retries exceeded on call
Hi: I have a asterisk server that shows the following Warning Jan 24 10:23:37 WARNING[1116941120]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Reque If it show on the midel of the call rhe call will be droped or if it show at the begining of the call the call will show buisy ( No one is available to answer at this time) Any help will be appreciate Please help Thanks Erick W ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Microsoft Portrait
Dose someone have been able to connect Micrrosofto Portrait Pocket PC Version to asterisk? Any information will be appreciate Thanks Erick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie setup (Hardware questions)
FXO is for connecting a line FXS is for connecting a Phone Best Regards Erick - Original Message - From: Puddle [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 15, 2004 4:59 PM Subject: [Asterisk-Users] Newbie setup (Hardware questions) Hello, I'm trying to setup an Asterix PBX solution in our office. We plan to have 5 active lines open available at any point in time. We'd like to use VoIP Phones, and possibly Software Based phone (*NIX/Windows enviroment). I was researching the various cards and I think I'd want to go with the Digium TDM40B - 4-port. However, I can't figure the differences between FXS FXO to make a decision of a purchase. Could someone help explain this? Ideally, we want to have one main line and have it hit a directory then branch to each work area with a phone. Thanks, -Puddle __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC and Pattern question
You have to use Regex There's a program call The Regex Coach that is very usefull Erick - Original Message - From: Eric Hall [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, November 30, 2004 9:20 AM Subject: [Asterisk-Users] ASTCC and Pattern question Hello group I just installed ASTCC and it was VERY easy to get running. I have a question about Pattern Via the web page I click the Routes link and everything makes sense to me but the pattern part. I tried _NXXNXX with the idea that everything would match this. Well it doesn't work... Does anyone have a good how-to? Thanks for all your help!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HOW TO ASTCC?
Can someone help me on how to use the ASTCC add-on for asterisk, I have instaled it and my proble is on the Routes section I don´t know what I should put in the Patter space Any help will be appreciate Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HOW TO ASTCC?
Darren: Thanks a lot for your help, it worked like a charm. I´m woundering if I can use 1NXXNXX* Correct my if I´m wrong you used 1403.* this means that it has to match 1403 then . for any nuber after that and the * is for? Thanks Erick - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, September 20, 2004 6:25 PM Subject: Re: [Asterisk-Users] HOW TO ASTCC? Here is an example: I will pick northern Alberta, in Canada. The country code is 1 and the area code is 403. Use 1403.* Darren Wiebe [EMAIL PROTECTED] Erick Weber V. wrote: Can someone help me on how to use the ASTCC add-on for asterisk, I have instaled it and my proble is on the Routes section I don´t know what I should put in the Patter space Any help will be appreciate Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC Help!!!
Thank you very much for your help, I think is working now Erick - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, September 10, 2004 8:27 PM Subject: Re: [Asterisk-Users] ASTCC Help!!! http://asterisk.gnuinter.net Have a look at this website and download the asterisk-perl package. That will solve the problem. Darren Wiebe Erick Weber V. wrote: I´m tring to install ASTCC for asterisk an I get the following error: mkdir -p /usr/local/apache/html/_astcc mkdir -p /usr/local/apache/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi /dev/null Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.0 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.0 /usr/lib/perl5/vendor_perl /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 .) at ./astcc.agi line 34. BEGIN failed--compilation aborted at ./astcc.agi line 34. make: *** [install] Error 2 Could somebody HELP me :) Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC Help!!!
I´m tring to install ASTCC for asterisk an I get the following error: mkdir -p /usr/local/apache/html/_astcc mkdir -p /usr/local/apache/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi /dev/null Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 /usr/lib/perl5/site_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.0 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.0/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.0 /usr/lib/perl5/vendor_perl /usr/lib/perl5/5.8.0/i386-linux-thread-multi /usr/lib/perl5/5.8.0 .) at ./astcc.agi line 34. BEGIN failed--compilation aborted at ./astcc.agi line 34. make: *** [install] Error 2 Could somebody HELP me :) Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] miserable time with Cisco ATA186
Yo have to stop and restart asterisk to get the new seting to work, not reload Erick - Original Message - From: Matthew Simpson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 04, 2004 8:52 AM Subject: Re: [Asterisk-Users] miserable time with Cisco ATA186 If I turn allow=ulaw on only, asterisk tries to use it a=rtpmap:0 PCMU/8000 but the ATA says it doesn't have it: Answering/Requesting with root capability 4 Answering with non-codec capability 0x1(G723) If I turn allow=alaw on only or with allow=ulaw, asterisk sends it, the ATA says it has it [alaw], but it still won't negotiate it. I think the stupid ATA is just determined to use G723 no matter what... I have LBRCodec set to 3 which should have it try to use G729, but it still tries to use G723. The AudioMode setting has a parameter bit to Enable G711 only, but I'm not sure how that bit thing works. Either the default 0x00150015 or the recommended 0x00140014 fails. [btw, bit 1 should be 1 to enable G711 only, if someone can help me there]. I'm seriously about to punt this thing into the garbage. Help! thanks, matt From: Timothy R. McKee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] miserable time with Cisco ATA186 Date: Fri, 4 Jun 2004 00:04:22 -0400 Reply-To: [EMAIL PROTECTED] Noticed that he has ALAW set as the preferred codec on the ATA. I'd suggest testing with allow of ulaw only, then try turning on other codecs. We know that one works well. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Thursday, June 03, 2004 23:36 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] miserable time with Cisco ATA186 Perhaps, but *I* at least had decent luck with 2.16.1. I suspect he has allow=all and the codec that ends up being used is G723.1 and then, of course, everything goes to hell. On Thu, 2004-06-03 at 22:59, brian k. west wrote: because 2.16.1 has some bugs.. you need 2.16.2 or higher. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] miserable time with Cisco ATA186
also you have to set the sip.conf on the ATA sittings to disallow=all, allow=g729 OR set the txcodec and rxcodec to 3 on the ATA - Original Message - From: Matthew Simpson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, June 04, 2004 8:52 AM Subject: Re: [Asterisk-Users] miserable time with Cisco ATA186 If I turn allow=ulaw on only, asterisk tries to use it a=rtpmap:0 PCMU/8000 but the ATA says it doesn't have it: Answering/Requesting with root capability 4 Answering with non-codec capability 0x1(G723) If I turn allow=alaw on only or with allow=ulaw, asterisk sends it, the ATA says it has it [alaw], but it still won't negotiate it. I think the stupid ATA is just determined to use G723 no matter what... I have LBRCodec set to 3 which should have it try to use G729, but it still tries to use G723. The AudioMode setting has a parameter bit to Enable G711 only, but I'm not sure how that bit thing works. Either the default 0x00150015 or the recommended 0x00140014 fails. [btw, bit 1 should be 1 to enable G711 only, if someone can help me there]. I'm seriously about to punt this thing into the garbage. Help! thanks, matt From: Timothy R. McKee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] miserable time with Cisco ATA186 Date: Fri, 4 Jun 2004 00:04:22 -0400 Reply-To: [EMAIL PROTECTED] Noticed that he has ALAW set as the preferred codec on the ATA. I'd suggest testing with allow of ulaw only, then try turning on other codecs. We know that one works well. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Thursday, June 03, 2004 23:36 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] miserable time with Cisco ATA186 Perhaps, but *I* at least had decent luck with 2.16.1. I suspect he has allow=all and the codec that ends up being used is G723.1 and then, of course, everything goes to hell. On Thu, 2004-06-03 at 22:59, brian k. west wrote: because 2.16.1 has some bugs.. you need 2.16.2 or higher. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX or SIP termination provider
I'm in Mexico an I'll like to know wish is the best IAX or SIP Termination provider. Im tring to start a small Pre-paid long distance service. Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXS = FXO Converter Problem
Hello: I have a ATA 186 and a FXS = FXO converter so I will like to program a extension that can be dialed and it will dial the ATA extention #, wait for dial tone and then dial the phone number. Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS = FXO Converter Problem
Andrew
Thanks for your answer
I'll test this conf an I'll post it so you know if it works
Thanks
Erick
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, April 13, 2004 12:48 PM
Subject: RE: [Asterisk-Users] FXS = FXO Converter Problem
Erick Weber V. wrote:
Hello:
I have a ATA 186 and a FXS = FXO converter so I will like to program
a extension that can be dialed and it will dial the ATA extention #,
wait for dial tone and then dial the phone number.
Unfortunately I don't believe there is a concept of wait for dial tone.
You'll just need to test it, and see how long it takes to do the answer,
pickup, get dialtone. Time it a few times, then add a second or so onto
that.
I tried to write up an example of what you should put into your
extensions.conf, but It's a little over my head in this case. My thoughts
are:
exten = _91NX,1,Dial(sip/yourata)
exten = _91NX,1,SendDigits(${EXTEN:1})
NOTE: I've not tested this...
-
Andrew Thompson
http://aktzero.com/
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Re: [Asterisk-Users] FXS = FXO Converter Problem
Andrew:
It didn't work, the problem is that * stays on priority 1 until you hangup
and the it pass to priority 2 so what I think is that it has to be all in
the priority 1 line
Hope we can figure it out
Erick
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, April 13, 2004 12:48 PM
Subject: RE: [Asterisk-Users] FXS = FXO Converter Problem
Erick Weber V. wrote:
Hello:
I have a ATA 186 and a FXS = FXO converter so I will like to program
a extension that can be dialed and it will dial the ATA extention #,
wait for dial tone and then dial the phone number.
Unfortunately I don't believe there is a concept of wait for dial tone.
You'll just need to test it, and see how long it takes to do the answer,
pickup, get dialtone. Time it a few times, then add a second or so onto
that.
I tried to write up an example of what you should put into your
extensions.conf, but It's a little over my head in this case. My thoughts
are:
exten = _91NX,1,Dial(sip/yourata)
exten = _91NX,1,SendDigits(${EXTEN:1})
NOTE: I've not tested this...
-
Andrew Thompson
http://aktzero.com/
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[Asterisk-Users] Insert pause in SIP String
Hello: I'll Like to now how to insert a pause on a SIP string. I have a ATA 186 and a FXS = FXO converter so I will like to program a extension that can be dialed and it will dial the ATA extention, wait for dial tone and then dial the phone number. Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA 182 and *
Hi to everyone: Does someone know if the ATA 182 works OK with asterisk or should I get a HandyTone 486 instade or an ATA 186 and a FXS to FXO converter Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * and IConnectHere
Hi to everyone When I dial a phone numer using my IConnectHere acount I get this message. Can someone tell me what it is? Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424
Re: [Asterisk-Users] Codec negotation with re-invites..
I think it's because in de [general] section you only allow=ulaw and you shold allow=g729 to. I'm a newbie, hope I can help Best Regards Erick - Original Message - From: Billy Huddleston [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 12, 2004 1:02 PM Subject: [Asterisk-Users] Codec negotation with re-invites.. I'm about over this.. okay,, here is what I got.. [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = inbound ; Default for incoming calls tos=lowdelay tos=184 disallow=all; Disallow all codecs allow=ulaw [gateway] type=friend host=1.1.6.9 canreinvite=yes qualify=yes dtmfmode=rfc2833 context=default disallow=all allow=ulaw allow=g729 [sipphoneg729] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance-g729 dtmfmode=rfc2833 mailbox=2199 disallow=all allow=g729 [sipphoneulaw] type=friend secret=password nat=yes host=dynamic canreinvite=yes qualify=200 context=longdistance dtmfmode=rfc2833 mailbox=2199 disallow=all allow=ulaw okay, when I place a call from sipphoneulaw to the outside world via gateway, everything works fine.. If I place a call from sipphoneg729, it doesn't work.. One leg to the gateway will be ulaw, the leg to the phone will be g729, and, I have 1 way audio.. The sip phone can hear anything from the gateway, but, the gateway can't hear the phone. I've even went as far as to setup a seperate context for the g729 phone and do this.. ,SetVar,SIP_CODEC=g729 which, says it sets it to g729, but it's still a ulaw call.. Guys, this is a real problem... We're going be doing mixed configs.. and if a gateway says it can do both, and phone says it can only do one... then we should be using the compatable codec... PLEASE help.. This is going to cause problems in our rollout. Thanks, Billy +--+ | Billy Huddleston Senior Systems Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best ATA 186 Firmware
Hi: Someone know wich is the best firmware for the ATA 186 with * Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA 186 Registration!!!!
I'm tring to register my ATA to * and I getting the following message: Feb 23 18:13:04 NOTICE[1125329600]: chan_sip.c:5405 handle_request: Registration from 'sip:[EMAIL PROTECTED] user=phone' failed for 'xxx.xxx.xxx.xxx' I don't know what's wrong an why it register as user=phone??? Coul some one help me Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA 186 Registration!!!!
Thank you very much I just make the change and I'm up an running. One more quick question, why I can not hear the ring in the phone connected to the ATA, My extensions are configure as follow: exten = 106,1,Dial(SIP/106,30,tr) Thanks for the quick response Best Regards Erick - Original Message - From: Ejay Hire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 23, 2004 7:01 PM Subject: RE: [Asterisk-Users] ATA 186 Registration Hi. Open http://ip.of.your.ata/dev and set LoginID0: and LoginID1: to your login Id. Then set UseLoginID: to 1. If you haven't already, set a password on the ATA by entering it into the UIPassword: field. This field does not have repeat to confirm, so type carefully. Last but not least, there is a bug in the v3.0 code for the 186's. If you use the TOS bit's to mark SIP for QOS, downgrade back to 2.1.6. In 3.0, the ata sets TOS to 0x0, and ignores the TOS: configuration field. Hope that helps, Ejay Hire ISDN-Net Network Engineer ...Providing VoIP services to Tennessee businesses since 2003 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Monday, February 23, 2004 5:24 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ATA 186 Registration I'm tring to register my ATA to * and I getting the following message: Feb 23 18:13:04 NOTICE[1125329600]: chan_sip.c:5405 handle_request: Registration from 'sip:[EMAIL PROTECTED] user=phone' failed for 'xxx.xxx.xxx.xxx' I don't know what's wrong an why it register as user=phone??? Coul some one help me Thanks Erick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1
I'm new with * I have a question, how do I update * Thanks Erick - Original Message - From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 9:50 AM Subject: [Asterisk-Users] Asterisk 0.7.1 Asterisk 0.7.1 has been released fixing a few minor bugs. Thanks again to the bug marshalls, especially Malcolm and bkw. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
