Re: [asterisk-users] choppy sound
Hardware echo usually helps. You can aslo try using OSLEC. - Original Message - From: B.Masoud @ SH To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Friday, October 09, 2009 23:50 Subject: Re: [asterisk-users] choppy sound Hi, I am using CentOS Asterisk 1.4 The server has 4GB RAM, 2Ghz Duo Core, and digium 24ports fxo no hardware echo cancelation Does hardware echo will help? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 09, 2009 11:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] choppy sound It would be helpful to know the OS, release of Asterisk, hardware, etc. In my case, I start getting excessive echoes at end of day, so I do a restart when convenient each morning around 4:00 AM. -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] choppy sound
Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] choppy sound
It would be helpful to know the OS, release of Asterisk, hardware, etc. In my case, I start getting excessive echoes at end of day, so I do a restart when convenient each morning around 4:00 AM. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] choppy sound
Hi, I am using CentOS Asterisk 1.4 The server has 4GB RAM, 2Ghz Duo Core, and digium 24ports fxo no hardware echo cancelation Does hardware echo will help? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 09, 2009 11:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] choppy sound It would be helpful to know the OS, release of Asterisk, hardware, etc. In my case, I start getting excessive echoes at end of day, so I do a restart when convenient each morning around 4:00 AM. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] choppy sound
By the way, how to schedule auto reboot? thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 09, 2009 11:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] choppy sound It would be helpful to know the OS, release of Asterisk, hardware, etc. In my case, I start getting excessive echoes at end of day, so I do a restart when convenient each morning around 4:00 AM. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choppy sound, SIP calls within LAN
Hi! I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE repository). As a clients I use XLite on Mac, all on the same LAN. Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM and plenty of disk space on LEVEL 5 RAID. Calls to another SIP server (also asterisk) hosted by another company are 100% OK, so it is clearly problem with my server setup. Background music (before pickup) runs fine, but transmitted voice sound is very choppy, no matter of which codec I use. I have searched over net, and implemented one by one every reasonable receipt found, including. highpriority = yes internal_timing = yes transmit_silence = no nat = yes localnet=192.168.0.0/255.255.0.0 externip = xx.xx.xx.xx dtmfmode=rfc2833 Downgrading asterisk did not solved problem, too. Anyone please help if possible.. Many thanks in advance for any suggestion(s). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy sound, SIP calls within LAN
On Fri, 2009-09-25 at 13:01 +0300, andreil1 wrote: Hi! I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE repository). As a clients I use XLite on Mac, all on the same LAN. Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM and plenty of disk space on LEVEL 5 RAID. Calls to another SIP server (also asterisk) hosted by another company are 100% OK, so it is clearly problem with my server setup. Background music (before pickup) runs fine, but transmitted voice sound is very choppy, no matter of which codec I use. I have searched over net, and implemented one by one every reasonable receipt found, including. highpriority = yes internal_timing = yes transmit_silence = no nat = yes localnet=192.168.0.0/255.255.0.0 externip = xx.xx.xx.xx dtmfmode=rfc2833 Downgrading asterisk did not solved problem, too. Anyone please help if possible.. Many thanks in advance for any suggestion(s). snip My first guess would be a network problem. Is there something different in the network path between the users and the hosted Asterisk server versus the users and the internal Asterisk server? Have you implement some form of CoS / QoS internally (one should)? If you run a continuous ping from a user to the internal Asterisk server, is there any packet loss or congestion (indicated by widely varying response times)? Just a few thoughts - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy Sound On Bridging From SIP-IAX
I had the same problem doing SIP - IAX, 1.4.19.1 as well as the last 1.4.22 In my case I was trying to do FAX and the blips were breaking lots of the faxes. My solution was to switch to T.38 over SIP and (cross my fingers) the problems haven't came back so far. I don't know the source of the problem, except that going to the latest (at that time) 1.4 did not solve it. I did switch to 1.6 because I wanted to try out the new app_fax and I've been very pleased with the results. But keep in mind I also dropped the SIP - IAX conversion at the same time, so I don't have a good data point. Now you have me curious, and I think I'll try doing the SIP - IAX conversion just to see if 1.6 made it any better. Maybe 1.6 would solve your problem too? On Sat, Jan 24, 2009 at 8:48 PM, Muiz Motani m...@askaritech.com wrote: I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX-IAX peers or SIP-SIP peers. My timing source is ztdummy. Does anybody have any ideas on the possible source of the problem? -- Muiz Motani m...@askaritech.com Askari Technologies ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choppy Sound On Bridging From SIP-IAX
I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX-IAX peers or SIP-SIP peers. My timing source is ztdummy. Does anybody have any ideas on the possible source of the problem? -- Muiz Motani m...@askaritech.com Askari Technologies ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy Sound On Bridging From SIP-IAX
Muiz Motani wrote: I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX-IAX peers or SIP-SIP peers. My timing source is ztdummy. Does anybody have any ideas on the possible source of the problem? All of my chopping problems where because of firewalls and a few of them I never figured out but they magically started working right. Can you check any firewall logs for dropped packets? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy sound while converting alaw to ulaw
Benoit Panizzon wrote: Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw = ulaw is choppy, ulaw = alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit- I do not believe that there is an issue in Asterisk. Is this a heavily used box? What does top show when making the call that is choppy. When you say your phone prefers it, do mean alaw is listed before ulaw? It that is the case, then it does not prefer it, it just came that way, default from the factory. Anyways, I find if bandwidth (and that is not even the case for ulaw/alaw) is a problem then transcode. If you do not have to transcode, use the same codec end to end. Then it is just passing data on the wire and not CPU intensive. Thanks, Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choppy sound while converting alaw to ulaw
Hi there I europe alaw is usual. I have a SIP Phone which perferes ulaw. When my * box has to transcode alaw to ulaw the sound get's one way choppy. (alaw = ulaw is choppy, ulaw = alaw is fine). I managed to fix the issue by forcing my SIP phone to use alaw only, but is this a know issue with asterisk 1.2.13? -Benoit- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] choppy sound when transcoding (after os update)
after recompilling asterisk (trunk-r75109) after system (mandriva cooker) update (new glibc 2.6, gcc 4.2.1), sound starts very choppy, when codec translation is performed, if translation isn't needed, it sounds OK any idea? until update, everything worked fine. I'm using ztdummy as clock source. during compile, I got lot of errors... ael_main.c: In function ‘ast_context_add_ignorepat2’: ael_main.c:306: warning: passing argument 1 of ‘create_name’ discards qualifiers from pointer target type ael_main.c: In function ‘ast_context_add_switch2’: ael_main.c:328: warning: passing argument 1 of ‘create_name’ discards qualifiers from pointer target type ael_main.c: In function ‘ast_context_add_include2’: ael_main.c:317: warning: passing argument 1 of ‘create_name’ discards qualifiers from pointer target type [CC] ast_expr2f.c - ast_expr2f.o ast_expr2.fl: In function ‘ast_yyerror’: ast_expr2.fl:376: warning: passing argument 1 of ‘expr2_token_subst’ discards qualifiers from pointer target type chan_agent.c: In function ‘__agent_start_monitoring’: chan_agent.c:393: warning: the address of ‘savecallsin’ will always evaluate as ‘true’ chan_agent.c:396: warning: the address of ‘urlprefix’ will always evaluate as ‘true’ [LD] chan_agent.o - chan_agent.so [CC] chan_iax2.c - chan_iax2.o chan_iax2.c: In function ‘iax2_prune_realtime’: chan_iax2.c:2050: warning: passing argument 1 of ‘expire_registry’ discards qualifiers from pointer target type ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] choppy sound with playback, background, etc... but not with musiconhold
Paco Brufal wrote: I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like tremolo or vibrato, but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()... Do you know what is happening and how can I fix it? It's an only SIP system, no fxo/fxs cards. Paco, You are in luck, because we just solved this problem. It turned out to be a timing issue, which surprised me because I thought timing was only important for IAX trunks, meetme rooms, and music-on-hold. Now I know that anything playing back audio files, except native music-on-hold, needs a reliable timing source. This includes the Playback() and Background() applications, as well as queue announcements. You have two options. One of them is free and might work and the other will cost about $75 per server but should work reliably. The first option is to use ztdummy, but there are a few points that you have to be aware of: 1. The kernel version must be at least 2.6.13 2. The kernel must be configured with a timer frequency of 1000 HZ 3. The kernel must be configured to provide RTC interrupts 4. The kernel must be configured with enhanced real time clock support With a properly configured 2.6.13 or greater kernel, ztdummy will use the RTC instead of kernel jiffies. This method is more accurate and should help alleviate your problems. Just remember that if you install a new kernel you'll have to rebuild Zaptel against it. We tried this and it helped, but it introduced another problem. The Playback() application would intermittently lock up. We are running on Dell PowerEdge 6850s, so I'm assuming the following issue documented at http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation was the cause: Mar 2006: Specifically, with many Dell servers, RTC will fail to give out interrupts, causing ztdummy to give no timing information. If you are using ztdummy, and having issues with the Playback() command causing the application to hang, then try the above step of disabling acpi. In addition, a recompiled kernel, with HPET_EMULATE_RTC option enabled, may solve the problem. (This option was removed in 2.6.13 kernel) The second option is to install a TDM400P without any FXS/FXO modules as a timing source. We've been running like this since Saturday and everything seems good so far. In this case, there are two points to keep in mind: 1. The server must have an available PCI slot that is compatible with the TDM400P 2. The wctdm module must be passed the 'timingonly=1' parameter at load Bare TDM400Ps are available from Atacomm at http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-45624523776.htm for under $75, which is pretty cheap for a just works solution. Remember that you can use zttest to verify the accuracy of your timing source. I ran it on each of the servers I installed a TDM400P in, and they all reported an average accuracy of better than 99.99%. Digium recommends an accuracy of at least 99.98%, which ztdummy using the RTC wasn't able to provide. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] choppy sound with playback, background, etc... but not with musiconhold
Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like tremolo or vibrato, but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()... If I move app_playback.so from this system to another asterisk, playback works fine... Do you know what is happening and how can I fix it? It's an only SIP system, no fxo/fxs cards. Thanks in advance. -- Paco Brufal[EMAIL PROTECTED] ServiTux Servicios Informáticos S.L. Tel. 966 160 600 / Fax. 966 160 601 ___ --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] choppy sound with playback, background, etc... but not with musiconhold
On Fri, 8 Jun 2007, Paco Brufal wrote: Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like tremolo or vibrato, but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()... If I move app_playback.so from this system to another asterisk, playback works fine... Do you know what is happening and how can I fix it? It's an only SIP system, no fxo/fxs cards. Do you have ztdummy loaded? Gordon ___ --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] Choppy sound with chan_capi + Fritz Card USB
Hi Everyone, Google IS my friend. I found the solution via Google on the second glance ;-) It seems that the USB latency was too high and you had to increase a CAPI-Buffersize in chan_capi.h: #define CAPI_MAX_B3_BLOCK_SIZE 500 (German instructions: http://www.ip-phone-forum.de/showthread.php?t=117614) Christoph Hi everybody, I have a problem which I cannot eliminate on my own. Has anybody any idea for the following: I am using the asterisk-version from Debian-Testing (1.2.13) with the latest chan_capi (also tried an older version). When using the Capi-Channel, everything works fine except from the sound it sounds extremely choppy and is unusable :-( When e.g. capisuite is used for fax, everything sounds fine... I found the following when using capi debug: ISDN1#02: too much voice to send for NCCI=0x10101 Google finds nothing relevant for this error message :-( Has anybody any idea ? Christoph ___ --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] Choppy sound with chan_capi + Fritz Card USB
Hi everybody, I have a problem which I cannot eliminate on my own. Has anybody any idea for the following: I am using the asterisk-version from Debian-Testing (1.2.13) with the latest chan_capi (also tried an older version). When using the Capi-Channel, everything works fine except from the sound it sounds extremely choppy and is unusable :-( When e.g. capisuite is used for fax, everything sounds fine... I found the following when using capi debug: ISDN1#02: too much voice to send for NCCI=0x10101 Google finds nothing relevant for this error message :-( Has anybody any idea ? Christoph P.S.: Here is the output of capi debug CONNECT_IND ID=002 #0x016e LEN=0037 Controller/PLCI/NCCI= 0x101 CIPValue= 0x10 CalledPartyNumber = c1XXX CallingPartyNumber = 00 a3 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo = default -- CONNECT_IND (PLCI=0x101,DID=XXX,CID=,CIP=0x10,CONTROLLER=0x1) ISDN1#02: msn='*' DNID='XXX' MSN == ISDN1#02: setting format alaw - 0x8 (alaw) == ISDN1#02: Incoming call '' - 'XXX' INFO_IND ID=002 #0x016f LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x1e InfoElement = 80 83 INFO_RESP ID=002 #0x016f LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1#02: info element PI 80 83 ISDN1#02: Origination is non ISDN INFO_IND ID=002 #0x0170 LEN=0022 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x70 InfoElement = c1XXX INFO_RESP ID=002 #0x0170 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1#02: info element CALLED PARTY NUMBER ISDN1#02: INFO_IND DID digits not used in this state. INFO_IND ID=002 #0x0171 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x18 InfoElement = 8a INFO_RESP ID=002 #0x0171 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1#02: info element CHANNEL IDENTIFICATION 8a INFO_IND ID=002 #0x0172 LEN=0016 Controller/PLCI/NCCI= 0x101 InfoNumber = 0xa1 InfoElement = a1 INFO_RESP ID=002 #0x0172 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1#02: info element Sending Complete -- ISDN1#02: CAPI/ISDN1/XXX-3: XXX matches in context external -- Executing VoiceMail(CAPI/ISDN1/XXX-3, 1234) in new stack == ISDN1#02: Answering for XXX CONNECT_RESP ID=002 #0x016e LEN=0042 Controller/PLCI/NCCI= 0x101 Reject = 0x0 BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default GlobalConfiguration= default ConnectedNumber = 00 80XXX ConnectedSubaddress = default LLC = default AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default SendingComplete= default -- Playing 'vm-intro' (language 'de') == Started pbx on channel CAPI/ISDN1/XXX-3 CAPI devicestate requested for ISDN1/XXX CAPI devicestate requested for ISDN1/XXX CONNECT_ACTIVE_IND ID=002 #0x0175 LEN=0015 Controller/PLCI/NCCI= 0x101 ConnectedNumber = default ConnectedSubaddress = default LLC = default CONNECT_ACTIVE_RESP ID=002 #0x0175 LEN=0012 Controller/PLCI/NCCI= 0x101 CONNECT_B3_IND ID=002 #0x0176 LEN=0013 Controller/PLCI/NCCI= 0x10101 NCPI= default CONNECT_B3_RESP ID=002 #0x0176 LEN=0015 Controller/PLCI/NCCI= 0x10101 Reject = 0x0 NCPI= default CONNECT_B3_ACTIVE_IND ID=002 #0x0177 LEN=0013 Controller/PLCI/NCCI= 0x10101 NCPI= default CONNECT_B3_ACTIVE_RESP ID=002 #0x0177 LEN=0012 Controller/PLCI/NCCI= 0x10101 DATA_B3_CONF ID=002 #0x0143 LEN=0016 Controller/PLCI/NCCI= 0x10101 DataHandle = 0x13a Info= 0x0 DATA_B3_REQ ID=002 #0x0143 LEN=0030 Controller/PLCI/NCCI= 0x10101 Data32 = 0x8168df4 DataLength = 0xa0 DataHandle = 0x13a Flags = 0x0 Data64 = 0x0 DATA_B3_REQ ID=002
[asterisk-users] Choppy sound in voicemail using Asterisk 1.2.11 on CENTOS4 guest on vmware server
I have had no success in getting the voicemail working on Asterisk 1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or without ztdummy device, renice -20 on asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when asterisk starts saying the digits from the extension, the sound starts becoming very choppy. The voice after the digits is still choppy. Does anyone have a suggestion? The codec that asterisk is using with the softphone I am using is the GSM codec. Please advise, Mario ___ --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] Choppy sound zap-to-sip, but not sip-to-sip?
Hi, I have lately noticed that we sometimes get choppy sound when recieving calls from the PSTN (on a TE410P-card) that get sent to an external SIP extension (over the internet) who has a somewhat bad connection. The strange thing is that it still sounds good when calling internally to the SIP-to-SIP. Is there any simple answer to why Zap-to-SIP (external) sounds bad when there is a bad connection, but SIP-to-SIP doesn't? The problem (I think) is not with the card or drivers since the problem only occurs when the connection is bad and never on our phones that are on the same internal network with the server. Thanks! Regards, Jan ___ --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
SV: [asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip?
Sorry. It sould say SIP-to-Zap not the other way around. Meaning that the Zap user is heard fine, but the external-SIP user is choppy when calling out on Zap (not when calling SIP-to-SIP though). -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 21 augusti 2006 15:15 Till: asterisk-users@lists.digium.com Ämne: [asterisk-users] Choppy sound zap-to-sip, but not sip-to-sip? Hi, I have lately noticed that we sometimes get choppy sound when recieving calls from the PSTN (on a TE410P-card) that get sent to an external SIP extension (over the internet) who has a somewhat bad connection. The strange thing is that it still sounds good when calling internally to the SIP-to-SIP. Is there any simple answer to why Zap-to-SIP (external) sounds bad when there is a bad connection, but SIP-to-SIP doesn't? The problem (I think) is not with the card or drivers since the problem only occurs when the connection is bad and never on our phones that are on the same internal network with the server. Thanks! Regards, Jan ___ --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
Re: [Asterisk-Users] Choppy Sound when using linux router or asterisk
While the gurus are sleeping, I'll ask a few questions to get started... I'm assuming this is using some kind of Voip, correct? If not, can you let us know what cards are involved? What kind of phone? Is it IP or analog? But, to be honest, none of that should hinder performance. What distribution of linux are you using? Do you have X loaded? That's not a good thing. If strictly Voip, is ztdummy loaded? I had some weird things happening on a VMware session that I think was related to timing (music on hold doing weird things). Is this plain vanilla asterisk, an rpm somewhere, asterisk at home, astlinux? Any other info would greatly be appreciated. In a typical situation, your hardware should be fine with no tweaks needed. That suggests something else is at fault. What speed is your network? Do you have a hub or a switch? Is this wireless or wired? What machine are you on doing the ssh to the router? How many other services are running on the linux machines? How many of systems are connected to the network? On 4/10/06, Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] wrote: Hello,I created this setup,DSL--LINUX ROUTER---ASTERISKLinux acts as router and forwards packets only 512M and AMD 1599.987 MHzAsterisk512MAMD 2000 MHzWhen I ssh to linux router during the call andexecute any command that requires cpu , then sound gets choppy.Simple test would be establish a call and start du / on the router. The same applies to asterisk box.Does anyone have any experience with tweaking the servers for bestperfomance with asterisk. How to give prority to asterisk processes , andto routing processes? Thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
Re: [Asterisk-Users] Choppy Sound when using linux router or asterisk
While the gurus are sleeping, I'll ask a few questions to get started... I'm assuming this is using some kind of Voip, correct? If not, can you let us know what cards are involved? What kind of phone? Is it IP or analog? I use sip protocol with broadvoice. But, to be honest, none of that should hinder performance. What distribution of linux are you using? Do you have X loaded? That's not a good thing. If strictly Voip, is ztdummy loaded? I had some weird things happening on a VMware session that I think was related to timing (music on hold doing weird things). Fedora core 4, no x Is this plain vanilla asterisk, an rpm somewhere, asterisk at home, astlinux? asterisk from ftp Any other info would greatly be appreciated. In a typical situation, your hardware should be fine with no tweaks needed. That suggests something else is at fault. What speed is your network? Do you have a hub or a switch? Is this wireless or wired? What machine are you on doing the ssh to the router? How many other services are running on the linux machines? How many of systems are connected to the network? 100MB I connect from wireless machine on the same network. On 4/10/06, Bartosz Wegrzyn - asterisk [EMAIL PROTECTED] wrote: Hello, I created this setup, DSL--LINUX ROUTER---ASTERISK Linux acts as router and forwards packets only 512M and AMD 1599.987 MHz Asterisk 512M AMD 2000 MHz When I ssh to linux router during the call and execute any command that requires cpu , then sound gets choppy. Simple test would be establish a call and start du / on the router. The same applies to asterisk box. Does anyone have any experience with tweaking the servers for best perfomance with asterisk. How to give prority to asterisk processes , and to routing processes? Thanks ___ --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 ___ --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] Choppy Sound when using linux router or asterisk
Hello, I created this setup, DSL--LINUX ROUTER---ASTERISK Linux acts as router and forwards packets only 512M and AMD 1599.987 MHz Asterisk 512M AMD 2000 MHz When I ssh to linux router during the call and execute any command that requires cpu , then sound gets choppy. Simple test would be establish a call and start du / on the router. The same applies to asterisk box. Does anyone have any experience with tweaking the servers for best perfomance with asterisk. How to give prority to asterisk processes , and to routing processes? Thanks ___ --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] Choppy Sound on PSTN End
Title: Choppy Sound on PSTN End Hi all, I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux. Our call routing is like this: SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice SIP account - PSTN Calls seem to work great from user to user. However, calls from a SJPhone user to the PSTN are not so great. The SJPhone user hears the person on the PSTN perfectly I mean, completely flawless. However, the user on the PSTN end hears choppy / jittery, extraneous clicks, etc. Here is the SJPhone config: Audio Compression: G.711 Driver buffer size: 20 msec Driver input queue length: 6 Driver output queue length: 4 RTP jitter queue length: 6 Silence Suppression: No DTMF Sending: RFC 2833 Signal Duration (ms): 270 RTP Payload type: 101 Signal volume: 10 Pause duration (ms): 100 And the sip extension config (in Asterisk Management Portal): Allow: blank Canreinvite: no Disallow: gsm Dtmfmode: rfc2833 Host: dynamic Nat: yes (some users are behind NAT) Qualify: no Any ideas on what to do to get rid of the choppiness? Thanks! Tim ___ 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] Choppy Sound on PSTN End
I have the same problem on a Dell 1850 with a TE410P and have been attempting to narrow it down. Interrupts don't seem to be a problem and I have two PRIs from two different suppliers and both have the same static/chop on the line so it's not the PRI. The leading suspect at the moment is the RAID controller. Unfortunately it's rather difficult to remove this from the set up but I plan to switch one of the PRIs to a Dell 1750 without a RAID controller to see if the problem still goes away. Aaron From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Chandler Sent: 02 May 2005 17:23 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Choppy Sound on PSTN End Hi all, I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux. Our call routing is like this: SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice SIP account - PSTN Calls seem to work great from user to user. However, calls from a SJPhone user to the PSTN are not so great. The SJPhone user hears the person on the PSTN perfectly I mean, completely flawless. However, the user on the PSTN end hears choppy / jittery, extraneous clicks, etc. Here is the SJPhone config: Audio Compression: G.711 Driver buffer size: 20 msec Driver input queue length: 6 Driver output queue length: 4 RTP jitter queue length: 6 Silence Suppression: No DTMF Sending: RFC 2833 Signal Duration (ms): 270 RTP Payload type: 101 Signal volume: 10 Pause duration (ms): 100 And the sip extension config (in Asterisk Management Portal): Allow: blank Canreinvite: no Disallow: gsm Dtmfmode: rfc2833 Host: dynamic Nat: yes (some users are behind NAT) Qualify: no Any ideas on what to do to get rid of the choppiness? Thanks! Tim ___ 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] Choppy Sound on PSTN End
Title: Choppy Sound on PSTN End I have the exact setup you describe, SJPhone - * - Zap/PRI. I think you need to twiddle some settings. You might turn on qualify just to see if the * is seeing network flaws. Keep in mind, if your using windows, anytime the user starts clicking around, you can expect less than ideal audio. Also, why disable GSM ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim ChandlerSent: Monday, May 02, 2005 11:23 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Choppy Sound on PSTN End Hi all, I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux. Our call routing is like this: SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice SIP account - PSTN Calls seem to work great from user to user. However, calls from a SJPhone user to the PSTN are not so great. The SJPhone user hears the person on the PSTN perfectly I mean, completely flawless. However, the user on the PSTN end hears choppy / jittery, extraneous clicks, etc. Here is the SJPhone config: Audio Compression: G.711 Driver buffer size: 20 msec Driver input queue length: 6 Driver output queue length: 4 RTP jitter queue length: 6 Silence Suppression: No DTMF Sending: RFC 2833 Signal Duration (ms): 270 RTP Payload type: 101 Signal volume: 10 Pause duration (ms): 100 And the sip extension config (in Asterisk Management Portal): Allow: blank Canreinvite: no Disallow: gsm Dtmfmode: rfc2833 Host: dynamic Nat: yes (some users are behind NAT) Qualify: no Any ideas on what to do to get rid of the choppiness? Thanks! Tim ___ 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] Choppy Sound on PSTN End
Hi, I have the same problem on a Dell 1850 with a TE410P, static/chop on calls to through the TE410P, and have been attempting to narrow it down for the last week. Interrupts don't seem to be a problem and I have two PRIs from two different suppliers and both have the same static/chop on the line so it's not the PRI. The leading suspect at the moment is the RAID controller. Unfortunately it's rather difficult to remove this from the set up but I plan to switch one of the PRIs to a Dell 1750 without a RAID controller to see if the problem still goes away. Aaron From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Chandler Sent: 02 May 2005 17:23 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Choppy Sound on PSTN End Hi all, I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux. Our call routing is like this: SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice SIP account - PSTN Calls seem to work great from user to user. However, calls from a SJPhone user to the PSTN are not so great. The SJPhone user hears the person on the PSTN perfectly I mean, completely flawless. However, the user on the PSTN end hears choppy / jittery, extraneous clicks, etc. Here is the SJPhone config: Audio Compression: G.711 Driver buffer size: 20 msec Driver input queue length: 6 Driver output queue length: 4 RTP jitter queue length: 6 Silence Suppression: No DTMF Sending: RFC 2833 Signal Duration (ms): 270 RTP Payload type: 101 Signal volume: 10 Pause duration (ms): 100 And the sip extension config (in Asterisk Management Portal): Allow: blank Canreinvite: no Disallow: gsm Dtmfmode: rfc2833 Host: dynamic Nat: yes (some users are behind NAT) Qualify: no Any ideas on what to do to get rid of the choppiness? Thanks! Tim ___ 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] Choppy Sound on PSTN End
I doubt it is the RAID controller since my Dell server isn't using one and I have this problem... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aza Sent: Monday, May 02, 2005 11:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Choppy Sound on PSTN End I have the same problem on a Dell 1850 with a TE410P and have been attempting to narrow it down. Interrupts don't seem to be a problem and I have two PRIs from two different suppliers and both have the same static/chop on the line so it's not the PRI. The leading suspect at the moment is the RAID controller. Unfortunately it's rather difficult to remove this from the set up but I plan to switch one of the PRIs to a Dell 1750 without a RAID controller to see if the problem still goes away. Aaron ___ 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] Choppy Sound on PSTN End
Title: Choppy Sound on PSTN End I turned on qualify=yes to see what happens. No effect that I can see. I have GSM disabled because I heard some bad things about the GSM protocol. I reenabled it, but to no avail. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Monday, May 02, 2005 11:22 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Choppy Sound on PSTN End I have the exact setup you describe, SJPhone - * - Zap/PRI. I think you need to twiddle some settings. You might turn on qualify just to see if the * is seeing network flaws. Keep in mind, if your using windows, anytime the user starts clicking around, you can expect less than ideal audio. Also, why disable GSM ? ___ 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] Choppy Sound on PSTN End
That's good to know about the RAID controller not being the problem, will save me some testing. I wonder if it's a change in asterisk then? Previously I was using E100Ps with asterisk-1.0.0 and didn't have the chop. Unfortunately at the same time I put in the TE410P I upgraded to asterisk-1.0.7 so don't know if the problem was introduced by the move from E100P to TE410P or from 1.0.0 to 1.0.7. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Chandler Sent: 02 May 2005 18:32 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Choppy Sound on PSTN End I doubt it is the RAID controller since my Dell server isn't using one and I have this problem... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aza Sent: Monday, May 02, 2005 11:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Choppy Sound on PSTN End I have the same problem on a Dell 1850 with a TE410P and have been attempting to narrow it down. Interrupts don't seem to be a problem and I have two PRIs from two different suppliers and both have the same static/chop on the line so it's not the PRI. The leading suspect at the moment is the RAID controller. Unfortunately it's rather difficult to remove this from the set up but I plan to switch one of the PRIs to a Dell 1750 without a RAID controller to see if the problem still goes away. Aaron ___ 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] choppy sound after 15 minutes in a call
I'm using X-Pro connected to an asterisk server (CVS-HEAD-01/27/05-23:17:07) and after about 15 minutes in a call I get a lot of noise in my end. I don't think the other part of the call hears it. After some 10 seconds or so everything is fine again. In my CLI I get NOTICE[32322]: RTP Transmission error to 85.xxx.xxx.xxx:35162: Operation not permitted. I get it on calls to the PSTN through my X100P (clone) as well as call connected through my IP telephony provider. I have also tried SJ-Phone and it happens with that as well. At the moment my asterisk server is on a public IP adress, and my client connects to it through an Intertex IX66 router. Before getting the router, I had dual NICs in the linux box and connected the client directly with the same problems... I've searched the lists but haven't been able to find any good answer, so any help is greatly appreciated :) /Anders ___ 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] choppy sound after 15 minutes in a call
On Tue, Feb 01, 2005 at 10:37:36PM +0100, Anders F Eriksson wrote: In my CLI I get NOTICE[32322]: RTP Transmission error to 85.xxx.xxx.xxx:35162: Operation not permitted. I get it on calls to the PSTN through my X100P (clone) as well as call connected through my IP telephony provider. I have also tried SJ-Phone and it happens with that as well. I don't know if you're using linux or not or exactly how the code is structured, but... if send() or write() to a network socket returns -EPERM it's generally that your firewall blocked it. Hope this helps, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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] Choppy sound ONLY when a voicemail is left
Hi All, Whenever a call comes in via the ISDN and somebody leaves a voicemail, the sound file recorded is very choppy. If I actually take the call, the sound is not choppy so it's obviously something to do with the Asterisk box itself having to do the recording. Perhaps the sound card drivers? I'm using the stock i810_audio (OSS) drivers on Fedora Core 1. If I call from a local VoIP client to the Asterisk box and leave a voicemail, the sound is NOT choppy. No errors on the Asterisk console. Thanks in advance. Regards, Gonzalo ___ 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
