RE: [Asterisk-Users] Asterisk with G729 codec does not want to connect with mediatrix SIP device
You need a G729 license for asterisk to make a connection. You have to get them from diguim, they are $10 a channel. They do give you a single channel demo license, you just have to get it from them. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Marko Rakar Sent: Thursday, March 25, 2004 8:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk with G729 codec does not want to connect with mediatrix SIP device I have tried to connect asterisk (which I use through hisax isdn4linux device) with mediatrix sip device with g729 codec asterisk can not connect with mediatrix (it connects when ulaw/alaw are used) when g729 is forced any ides what to do? Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9 Content-Length: 178 Content-Type: application/sdp Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, REFER v=0 o=MxSIP 0 0 IN IP4 192.168.3.211 s=SIP Call c=IN IP4 192.168.3.211 t=0 0 m=audio 5004 RTP/AVP 18 8 0 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 10 headers, 9 lines Found audio format UNKN Found audio format ALAW Found audio format UNKN Found description format G729 Found description format PCMA Found description format PCMU Capabilities: us - 268, them - 268/0, combined - 268 Non-codec capabilities: us - 1, them - 0, combined - 0 list_route: hop: sip:[EMAIL PROTECTED] set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9 From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.211:5060 Mar 25 15:24:26 NOTICE[1225991360]: channel.c:1513 ast_set_write_format: Unable to find a path from UNKN to SLINR Mar 25 15:24:26 WARNING[1225991360]: channel.c:1920 ast_channel_make_compatible: No path to translate from Modem[i4l]/ttyI0(64) to SIP/301-3309(256) Mar 25 15:24:26 WARNING[1225991360]: app_dial.c:702 dial_exec: Had to drop call because I couldn't make Modem[i4l]/ttyI0 compatible with SIP/301-3309 set_destination: Parsing sip:[EMAIL PROTECTED] for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK247473b9 From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.3.211:5060 asterisk*CLI Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE From: 0 sip:[EMAIL PROTECTED];tag=as01323dfd To: sip:[EMAIL PROTECTED];tag=674991B479A02CF7-370B96C56CAF118 Via: SIP/2.0/UDP 192.168.3.6;branch=z9hG4bK247473b9 Content-Length: 0 7 headers, 0 lines Sometimes you're the bug, sometimes you're the windshield. mailto:[EMAIL PROTECTED] http://printel.hr ___ 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] AGI test script
Did you make the file executable. chmod 777 *, what was the message given from konsole? Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Vikram Rangnekar Sent: Tuesday, March 16, 2004 2:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] AGI test script exten = 666,1,Answer exten = 666,2,AGI(agi-text.agi) exten = 666,103,Hangup iwhy is that not working any idea. Does answer need to be there or does the AGI script answer the call. -- regards Vikram (http://www.vicramresearch.com) ___ 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] chan_sip.c Error
What kind of device is at [1982]? I have gotten error messages like that when device at the specific IP is offline. Is a connection being made at all? Wes -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of [EMAIL PROTECTED]Sent: Tuesday, March 09, 2004 10:27 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] chan_sip.c Error Hello, I have implemented Asterisk on a private network (192.168.X.X/24). I've configured SIP peers and when I make a call between them, I've got these error messages in the * CLI and the communication stops by itself after 30 sec. WARNING[1142106560]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (request) WARNING[1142106560]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (request) Anyone can tell me why? I haven't found real solutions on the mails archive... Thanks you! - In sip.conf: [1982] type=friend username=user1 mailbox=1000 host=192.168.111.2 context=default In extensions.conf ... exten = 1982,1,Macro(std,SIP/1982) ... -Ë^®+$RÇ«²f¢)à+-Ë^®+$RÇ«²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®ÄèPÔ (®ê]jר¦Ø¨Ëâ²+a¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë
RE: [Asterisk-Users] G.729 vs. G.729 pass thru
I've had some small problems when trying to users features like AbsoluteTimeout with pass thru. Other than that sound quality has been good. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Unavailable ID Sent: Thursday, March 04, 2004 9:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru Darek, Thank you for the info. How is the sound quality when you are using with G.729 codec? What's your thought? Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 2:58 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru As long as you have a IDE drive available, and mounted when you install it, it will work. This includes CD ROM's...It's what I did. Funkiness with the registration process. As far as pass through goes, from what I understand, it *should*, but when you have licensed binaries, from what I've seen, it doesn't. It's actually used 2 licenses. I plan on figuring that out next. Derek From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729 vs. G.729 pass thru Hello everyone, If you don't have Digium card but you want to use G.729 codec, do you need a license for it? If the VoIP termination point supports G.729 and you are using sip phone (soft/hard phone), can you use the G.729 pass thru or you have to buy the license? Have anyone test it with SCSI system? Seems like it only work on machine with IDE disk. Thanks. ___ 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] G.729 vs. G.729 pass thru
Yes, I do something like that. MediatrixFXO(1204)-Asterisk-MediatrixFXO(1204), I have bought license from diguim for G.729. I do not really have a telco provider just an ISP. I use if for a private network. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Unavailable ID Sent: Friday, March 05, 2004 3:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru Hi Wes, Do you need to buy license when you are using pass thru. How does it work? I'm thinking about using pass thru for voip since the service provider has g.279 codec. Can you setup your * box connects to telco termination with pass thru? PBX =[T100P]= ASTERISK (*) =[G.729]= VOIP TERMINATION Thanks. - Original Message - From: Wes Marderness [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 05, 2004 11:42 AM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru I've had some small problems when trying to users features like AbsoluteTimeout with pass thru. Other than that sound quality has been good. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Unavailable ID Sent: Thursday, March 04, 2004 9:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G.729 vs. G.729 pass thru Darek, Thank you for the info. How is the sound quality when you are using with G.729 codec? What's your thought? Thanks. - Original Message - From: Derek Samford [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Thursday, March 04, 2004 2:58 PM Subject: RE: [Asterisk-Users] G.729 vs. G.729 pass thru As long as you have a IDE drive available, and mounted when you install it, it will work. This includes CD ROM's...It's what I did. Funkiness with the registration process. As far as pass through goes, from what I understand, it *should*, but when you have licensed binaries, from what I've seen, it doesn't. It's actually used 2 licenses. I plan on figuring that out next. Derek From: Unavailable ID [mailto:[EMAIL PROTECTED] Sent: Thursday, March 04, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G.729 vs. G.729 pass thru Hello everyone, If you don't have Digium card but you want to use G.729 codec, do you need a license for it? If the VoIP termination point supports G.729 and you are using sip phone (soft/hard phone), can you use the G.729 pass thru or you have to buy the license? Have anyone test it with SCSI system? Seems like it only work on machine with IDE disk. Thanks. ___ 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 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] add g.729 license
You can purchase more licenses from diguim. Asterisk will overwrite your previous license. You would have to get someone from diguim to combine your 2 different licenses into one. Like if you had 2 license and want to upgrade to 4. You would have to purchase another 2 channel license and have both the license combined. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ron McMillin Sent: Tuesday, March 02, 2004 1:58 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] add g.729 license Hi, I already have one g.729 license on *. Could anyone tell me if I want to add a few more, can I just buy these online and follow their installation instruction, and * will add these addtional licenses? Or this will invalidate my current license? thanks ron ___ 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] G729 troubles
I had a problem something like what you described. I was able to run asterisk from /tmp, had to bug digium for an answer for that one. Hope this helps wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Darren Wiebe Sent: Saturday, February 28, 2004 2:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G729 troubles I forgot to mention what I have been trying to fix it. I'm running it from the console asterisk -vvvcng but this does not help. I've searched the mailing lists and found a lot of messages with people having the same problem. I'll try calling digium Monday if I cannot resolve it today and see if they can help me. Darren Wiebe [EMAIL PROTECTED] Darren Wiebe wrote: I am a new asterisk user. I have had a box up and running for a couple of months and been very happy with it. Last night I came up with a question that I have not been able to find an answer too. I purchased 5 licenses for the G729 codec from digium. My source is current from CVS as of late last night. Here are messages I'm getting from Asterisk. Can anybody help me? [codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator) Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call == Detected 5 licensed G.729 transcoders Feb 28 08:47:49 WARNING[-1084456832]: translate.c:219 calc_cost: Translator 'g72 9tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 == Registered translator 'lintog729b' from format SLINR to G729A, cost 26 Thanks so much, Darren Wiebe [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 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] G729 troubles
My server is running fine now. I have to 'cd /tmp' then '/usr/sbin/asterisk -gc' or I receive error messages. It is very strange but it works. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Darren Wiebe Sent: Monday, March 01, 2004 11:23 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G729 troubles I have gotten in contact with Digium. As soon as we have something resolved, I will post the fix to the list. Thanks, Darren Wiebe [EMAIL PROTECTED] Brent Franks wrote: Wes, Please let us know how you make out with this. I experience the same issues. - Brent -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wes Marderness Sent: Monday, March 01, 2004 9:24 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] G729 troubles I had a problem something like what you described. I was able to run asterisk from /tmp, had to bug digium for an answer for that one. Hope this helps wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Darren Wiebe Sent: Saturday, February 28, 2004 2:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] G729 troubles I forgot to mention what I have been trying to fix it. I'm running it from the console asterisk -vvvcng but this does not help. I've searched the mailing lists and found a lot of messages with people having the same problem. I'll try calling digium Monday if I cannot resolve it today and see if they can help me. Darren Wiebe [EMAIL PROTECTED] Darren Wiebe wrote: I am a new asterisk user. I have had a box up and running for a couple of months and been very happy with it. Last night I came up with a question that I have not been able to find an answer too. I purchased 5 licenses for the G729 codec from digium. My source is current from CVS as of late last night. Here are messages I'm getting from Asterisk. Can anybody help me? [codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator) Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call Feb 28 08:47:49 WARNING[-1084458064]: asterisk.c:260 listener: Select retured er ror: Interrupted system call == Detected 5 licensed G.729 transcoders Feb 28 08:47:49 WARNING[-1084456832]: translate.c:219 calc_cost: Translator 'g72 9tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 == Registered translator 'lintog729b' from format SLINR to G729A, cost 26 Thanks so much, Darren Wiebe [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 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 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] SIP Re-Invites Timeout
Hi All, I have 2 FXO Gateways devices. Both under sip.conf are set to canreinvite=yes. My dailplan is below. exten = _X.,1,Absolutetimeout(40) exten = _X.,2,dial(SIP/[EMAIL PROTECTED]) exten = T,1,BackGround(tt-weasels) exten = T,2,Hangup() With this setup the absolute timeout is never triggered. I found that only when canreinvite=no does the absolute timeout fire. Also when canreinvite=no there is noticeable background noise on all calls. Used CVS version of asterisk dated today(02/24/04) for the test. Is what I am trying to do possible? Will SIP Absolute timeout work with Reinvites? Thanks ahead of time for anyone's help, Wesley Marderness Ph: 1-610-372-9092 ___ 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] g729 license
When you start * from console use -vvvc and the number of detected licenses will be shown when the g729 translator is loaded. Only why that I know of to check this. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Osvaldo Mundim Sent: Monday, February 16, 2004 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] g729 license Hello all, I wanted to know if is there a way to see which of my 4 g729b license is registered in one specific Asterisk box. Is that possible? I could not find any registration record on my box to compare with the license... best regards Osvaldo ___ 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] g729 license
I know that during the Registration that the file /var/lib/va-certificate is created. Maybe this will help, file is encrypted so it don't offer much information. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Osvaldo Mundim Sent: Monday, February 16, 2004 11:03 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] g729 license The problem is that I have 2 licenses of 8 channels. One is being used in one of my boxes and the other one is not. What I want is to be sure that the one which I will use in a new Asterisk box is not the one which is being used... Any suggestion? regards Osvaldo On Feb 16, 2004, at 11:57 AM, Wes Marderness wrote: When you start * from console use -vvvc and the number of detected licenses will be shown when the g729 translator is loaded. Only why that I know of to check this. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Osvaldo Mundim Sent: Monday, February 16, 2004 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] g729 license Hello all, I wanted to know if is there a way to see which of my 4 g729b license is registered in one specific Asterisk box. Is that possible? I could not find any registration record on my box to compare with the license... best regards Osvaldo ___ 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 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] SIP - NATIVE BRIDGE ERROR
I did 'sip show channels' during call. I saw 2 channels, one for each device, that where on G729A. I thought that when a Native bridge was done * was releasing the call. I did not think a call would require 2 G729 Channels because * is just initializing the session between the 2 devices. I have done the exact same thing on 0.5.0 with the demo license and did not receive any error or background noise. thanks for any help, Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian West Sent: Friday, February 06, 2004 7:39 PM To: Asterisk-Users Subject: Re: [Asterisk-Users] SIP - NATIVE BRIDGE ERROR Isn't the demo codec 1 channel only? Then one side is g729 and the other is what? do a sip show channels bkw On Fri, 6 Feb 2004, Wes Marderness wrote: Hi, Running Version 0.7.2, I receive the following error when attempting to connect two SIP Devices. WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1 = 524558, cannot native bridge. The bridge is made but the quality of the call is bad, a lot of disturbing noises in background. Oddly enough, both devices are using only one codec G729. I also am using the demo G729 license for Asterisk. I'm not sure how 2 different codecs are being found. I saw in ast_rtp_bridge function, that the get_codec function returned these values. Could anyone tell me where the get_codec function is? Curious as to how this is happening. Should this problem be added to the bug tracker? The SIP calls are very bad, and I did not experience this problem with 0.5.0 . Thanks, Wes ___ 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] Help with Sip call problems - Whats not working?
What does your extensions.conf look like? Did you answer() the call first ? wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Lee Sent: Monday, February 09, 2004 6:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help with Sip call problems - Whats not working? When I press a key (8) on the phone, it should play a few bits of audio and go to voicemail for testing. I dont get any sound back, and it appears the call is progressing without me. Here is the console output with sip debug: Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED] Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: 287 v=0 o=p3000 5972727 56415 IN IP4 10.10.10.2 s=SIP Call c=IN IP4 10.10.10.2 t=0 0 m=audio 10096 RTP/AVP 0 8 18 4 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 g723/8000 a=rtpmap:2 g726/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 13 lines Using latest request as basis request Sending to 10.10.10.2 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format ULAW Found audio format GSM Found audio format UNKN Found description format pcmu Found description format pcma Found description format G729 Found description format g723 Found description format g726 Found description format telephone-event Capabilities: us - 14, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 8 in wellingborough-road list_route: hop: sip:[EMAIL PROTECTED] Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 To: sip:[EMAIL PROTECTED];tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 10.10.10.2:5060 -- Executing BackGround(SIP/p3000-1186, sounds/carried-away-by-monkeys) in new stack We're at 10.10.10.3 port 17190 Answering with preferred capability 2 Answering with preferred capability 4 Answering with preferred capability 8 Answering with non-codec capability 1 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 To: sip:[EMAIL PROTECTED];tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17878 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to 10.10.10.2:5060 -- Playing 'sounds/carried-away-by-monkeys' (language 'en') babybell*CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED] Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: 287 v=0 o=p3000 5972727 56415 IN IP4 10.10.10.2 s=SIP Call c=IN IP4 10.10.10.2 t=0 0 m=audio 10096 RTP/AVP 0 8 18 4 2 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 g723/8000 a=rtpmap:2 g726/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 13 lines Ignoring this request We're at 10.10.10.3 port 17190 Answering with preferred capability 2 Answering with preferred capability 4 Answering with preferred capability 8 Answering with non-codec capability 1 Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-ZgX-11841 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 To: sip:[EMAIL PROTECTED];tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 232 v=0 o=root 17878 17879 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 17190 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 to 10.10.10.2:5060 babybell*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.2:5060;branch=z9hG4bK-AAE-26994 From: p3000 sip:[EMAIL PROTECTED]:5060;tag=TdR-16808 To: sip:[EMAIL PROTECTED];tag=as3bf9fee8 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK
[Asterisk-Users] SIP - Native Bridge Error
Hi, Running Version 0.7.2, I receive the following error when attempting to connect two SIP Devices. WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1 = 524558, cannot native bridge. The bridge is made but the quality of the call is bad, a lot of disturbing noises in background. Oddly enough, both devices are using only one codec G729. I also am using the demo G729 license for Asterisk. I'm not sure how 2 different codecs are being found. I saw in ast_rtp_bridge function, that the get_codec function returned these values. Could anyone tell me where the get_codec function is? Curious as to how this is happening. Should this problem be added to the bug tracker? The SIP calls are very bad, and I did not experience this problem with 0.5.0 . Thanks, Wes ___ 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] SIP - NATIVE BRIDGE ERROR
Hi, Running Version 0.7.2, I receive the following error when attempting to connect two SIP Devices. WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1 = 524558, cannot native bridge. The bridge is made but the quality of the call is bad, a lot of disturbing noises in background. Oddly enough, both devices are using only one codec G729. I also am using the demo G729 license for Asterisk. I'm not sure how 2 different codecs are being found. I saw in ast_rtp_bridge function, that the get_codec function returned these values. Could anyone tell me where the get_codec function is? Curious as to how this is happening. Should this problem be added to the bug tracker? The SIP calls are very bad, and I did not experience this problem with 0.5.0 . Thanks, Wes ___ 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] G729 license
I purchased a license from Digium, If you ask they will can also give you a trial license to test out. Wes -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Jess MagnayeSent: Friday, January 30, 2004 10:29 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] G729 licenseImportance: High Hello all, I would like to just verify where to purchase the G729 license for Asterisk. Like I want to run G729 codec for all my calls passing thru Asterisk (voicemail, parking, via ZAP, via SIP, etc). The list says license is taken from Digium, does that apply also if I have Dialogic cards on my *?
[Asterisk-Users] SIP Absolute Timeout
Hi All, I've been having a hard time getting the AbsoluteTimeout function to work. Is this Function working in for SIP? I've search all the messages in the news letters and tried what was suggested and still have not gotten it to work. Below is a portion of my extensions.conf. I've also been running these test on ver 0.5.0 exten = _X.,1,Absolutetimeout(20) exten = _X.,2,dial(SIP/[EMAIL PROTECTED]) exten = T,1,BackGround(tt-weasels) exten = T,2,Hangup() Thanks ahead of time for any help / suggestions. Wes Marderness ___ 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] Mediatrix 1204 sip experience?
UI for switch config allows you to generate scripts for setting if you need them. I found that to be useful. They can be easily configured from remote if you have the UI software. There are features for caller id, but I have not used them yet. Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Friday, January 23, 2004 2:40 PM To: Asterisk-a-users-list Subject: [Asterisk-Users] Mediatrix 1204 sip experience? Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich ___ 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] SIP AbsoluteTimeout
Hi, I've been having a hard time getting the absolutetimeout feature to work. I've search all the messages in the news letters and tried what was suggested and still have not gotten it to work. Below is a portion of my extensions.conf and sip.conf. I've also been running these test on ver 0.5.0 exten = _X.,1,Absolutetimeout(20) exten = _X.,2,dial(SIP/[EMAIL PROTECTED]) ;exten = _X.,2,AGI(callout-agi.agi) exten = T,1,BackGround(tt-weasels) exten = T,2,Hangup() ;sip.conf [general] port = 5060 bindaddr = 0.0.0.0 context = default allow=g729 maxexpirey= 60 defaultexpirey = 50 [SIPOUT#1] type=friend username=sipout secret=notreally context = default host=192.168.1.10 reinvite=yes disallow=all allow=g729 Thanks ahead of time for any help / suggestions. Wes Marderness ___ 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] Pls confirm
allow=g723.1 is valid but depending on the hardware you are using it might not passthru. make sure you set: 'reinvites=yes' to allow pass through, this does not always work on all devices. What features do you mean? I used the licensed G729 Wes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jess Magnaye Sent: Tuesday, January 06, 2004 3:24 PM To: [EMAIL PROTECTED] Subject: Fw: [Asterisk-Users] Pls confirm Importance: High - Original Message - From: Jess Magnaye [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm Is the format allow=g723.1 in sip.conf valid? somehow i cannot get it working to do g723 passthru. also, i've read that doing g723 will disable voicemail, and other * features. is that true? if it is, will the licensed-g729 work for * features? thanks again in advance. - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 1:38 PM Subject: Re: [Asterisk-Users] Pls confirm Jess Magnaye wrote: Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? It can do G.723 between endpoints (passthrough).. It can do G.729a with the purchase of an additional licence of $10 per channel.. Yes you can use G.711 with a provider, some providers offer GSM, iLBC and Speex as alternative codecs.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
