Re: [asterisk-users] Heavy duty environment - Is TDM2400P suits?
I am using TDM2400 with FXO modules to handle 16 concurrent calls to PSTN for more than 12 hours a day with no problem at all. On 8/16/07, Chan Jason [EMAIL PROTECTED] wrote: Hi all, I am planning to have a new TDM2400P to replace all Planet 450 SIP gateways. Can TDM2400P survive in heavy duty environment where there will be 4 concurrent calls in within the same second? Thanks! Yours sincerely, Jason Chan ___ --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
Re: [asterisk-users] Codec Negotiation
I do not need g723.1 codec, this is not the problem, here is another description of the problem: The client offer 2 codecs (g729 and g723) for all calls, my server accept only g729, so normally the client server will negotiate the codec and both sides agrees on g729, but this does not happened always, sometimes for some reason, my server reject the call giving a codec error below, which means that both sides did not nogotiate the codec correctly. On 7/12/07, Jared Smith [EMAIL PROTECTED] wrote: On Thu, 2007-07-12 at 14:39 -0400, Al Bochter wrote: So who do you pay to use the G723 codec? It's possible to use the G.723.1 codec with Asterisk by buying a Digium TC400B transcoder card[1]. Without that card, the best Asterisk can do is to pass through the packets, but it can't doing any transcoding to/from G.723.1 without the hardware card. --- Jared Smith Community Relations Manager Digium, Inc. [1] http://www.digium.com/en/products/hardware/tc400b.php ___ --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] Codec Negotiation
I am having a problem with my asterisk gateway, it is accepting only G729, the client is offering G729 and G723.1, however for some reasons, around 15% of calls are rejected due to failed codec negotiation giving an codec error No compatible codecs, not accepting this offer. Anyone gone through this before? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM800P cards with one way voice
I have 2 servers trunked IAX, one of them has 2 TDM800P cards to terminate calls to PSTN. the problem is all calls to PSTN is almost one-way voice. the voice is always broken. ztmonitor shows that most of the time of the call the Tx is zero, while there is always Rx activity on this channel. Any clue? I am using Asterisk 1.4.5, G729, SIP from clients to 1st server, IAX between servers, one of the server is NATed. ___ --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] connecting 2 asterisk servers through OpenVPN
There is no problem with the CPU utilization, it is around 40%, I will not be able to try this without the VPN, maybe I should try another VPN solution like OpenSwan, or PPTP. Why do you think that IAX will make a difference than SIP? On 1/16/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 15 Jan 2007, O.Kamal wrote: I am trying to connect 2 asterisk servers through OpenVPN, the VPN should carry 16 channel, however when active channels reached 4 concurrent channels, the connection became unstable, with a very high latency (around 900ms), the internet bandwidth is 1Mbps on each server, I have upgraded the bandwidth to double it, but still have exactly the same problem. Any tips or recommendations on such setup? No real answers, but questions that might help ... Have you tried it without using OpenVPN? Just port-forward the SIP RTP ports, if you need to and give it a go. I am using SIP and G729 between the 2 servers, openVPN using UDP with no compression. Why not IAX? Are your openVPN end-points up to it? Doing high-grade encryption in software might challenge some slower processors - are the VPN endpoints the asterisk boxes themselves? 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 ___ --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] connecting 2 asterisk servers through OpenVPN
I am trying to connect 2 asterisk servers through OpenVPN, the VPN should carry 16 channel, however when active channels reached 4 concurrent channels, the connection became unstable, with a very high latency (around 900ms), the internet bandwidth is 1Mbps on each server, I have upgraded the bandwidth to double it, but still have exactly the same problem. Any tips or recommendations on such setup? I am using SIP and G729 between the 2 servers, openVPN using UDP with no compression. Thanks, O.Youssef ___ --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] idle SIP channels problem
I have 2 asterisk servers connected together on internet, when placing one or two calls, things goes fine, but when placing more calls, i am getting the below messages on the far end: Jan 5 17:25:00 ERROR[2679] chan_sip.c: Call from peer 'switch' rejected due to usage limit of 16 Jan 5 17:25:00 NOTICE[2679] chan_sip.c: Failed to place call for user , too many calls I am seeing a lot of idle connection when doing sip show channels. ___ --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] postgres and asterisk
I need to retrieve my asterisk to retrieve a values from postgresql, i am looking for some sort of application like *mysql*() app, I found one but it is only available on Suse, is there any way for doing this? Regards, O.Youssef ___ --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] postgres and asterisk
I just need to retrieve a value from a field in a postgres database, and playback this value when someone dial a specific extension. On 1/4/07, Thomas Kenyon [EMAIL PROTECTED] wrote: O.Kamal wrote: I need to retrieve my asterisk to retrieve a values from postgresql, i am looking for some sort of application like *mysql*() app, I found one but it is only available on Suse, is there any way for doing this? Regards, O.Youssef What do you need to do? To get an SQL console with postgres you need to: psql -d database name to start in -U username to connect as ie: psql -d asterisk -U asterisk The location of psql is different depensing upon distribution but usually it's in either /usr/bin/psql or /usr/local/pgsql/bin/psql. I'm not sure if this is what you want, if you want a pretty GUI front-end then you could look at Pgadmin III (www.pgadmin.org) which will run on Windows 2000/XP/2003 or unix/linux running X and requires wxWindows and a pile of common libraries. ___ --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] ZAP problem
when placing calls to the system through SIP, I got these messages, Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator path exists for channel type Zap (native 68) to 256 Dec 19 00:26:55 NOTICE[5570]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) any explanation for this? 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
Re: [asterisk-users] ZAP problem
Why do I need g729 license?, i am not doing any transcoding in the middle. it is all g729 passthrough. softphone---asterisk---zap On 12/18/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 18, 2006 at 11:22:13PM +0200, O.Kamal wrote: when placing calls to the system through SIP, I got these messages, Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator path exists for channel type Zap (native 68) to 256 Dec 19 00:26:55 NOTICE[5570]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) boomtime*CLI show audio codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPENAME DESC 1 (1 0) (0x1) audiog723 (G.723.1) 2 (1 1) (0x2) audio gsm (GSM) 4 (1 2) (0x4) audioulaw (G.711 u-law) 8 (1 3) (0x8) audioalaw (G.711 A-law) 16 (1 4) (0x10) audiog726 (G.726) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audioslin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audiog729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audioilbc (iLBC) 68 is 64 + 4, that is: only the bits for ulaw and slinear are set. 256 means that only g729 is supported. Your system cannot transcode g729 to ulaw: you don't have a g729 codec installed, probably. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] ZAP problem
i have digium TDM2404E, I was thinking that zap devices are not related to any kind of codecs. I will try setting my soft phone and asterisk server to use ulaw, to see how things will go... On 12/18/06, Mailing List [EMAIL PROTECTED] wrote: What zap device do you have that encodes/decodes g729? - Original Message - *From:* O.Kamal [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Monday, December 18, 2006 4:37 PM *Subject:* Re: [asterisk-users] ZAP problem Why do I need g729 license?, i am not doing any transcoding in the middle. it is all g729 passthrough. softphone---asterisk---zap ___ --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] TDM2400
I am having 2 more issues, when starting asterisk I got the below message:
Dec 17 22:27:54 NOTICE[4554]: indications.c:505
ast_unregister_indication_country: Removed default indication country 'us'
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
signalling
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
answeronpolarityswitch
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
hanguponpolarityswitch
my setup is : softphone---softswitch(asterisk)Termination GW(asterisk
with TDM card)
when dialing from my softphone I got :
-- Executing Dial(SIP/10.8.0.6-0947d008, (Zap/g1/6159587)) in new
stack
Dec 17 23:04:16 WARNING[5052]: channel.c:2597 ast_request: No channel type
registered for '(Zap'
Dec 17 23:04:16 NOTICE[5052]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type '(Zap' (cause 66 - Channel not implemented)
my extensions.conf file has:
[globals]
TRUNK=Zap/g1
[topstn]
exten=_2.,1,Dial,(${TRUNK}/${EXTEN:3})
Please help ...
Thanks,
On 12/12/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Dec 11, 2006 at 08:08:18PM -0500, Time Bandit wrote:
[channels]
context=default
signalling=fxs_ls
;channel=1-16
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
;accountcode=lss0101
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
To the best of my knowledge, all the settings you put after defining
the channles (channel= line) are useless. You have to set all the
settings BEFORE you define the channels.
Should be. However in practice after the first reload all of them will
be applied (in this specific case).
/me points again to genzaptelconf that should have made this thread
unnecessary.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] TDM2400
After restarting the machine I am getting the below messages when dialing:
Dec 18 00:09:35 WARNING[2897]: channel.c:2571 ast_request: No translator
path exists for channel type Zap (native 68) to 256
Dec 18 00:09:35 NOTICE[2897]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0 - Unknown)
On 12/17/06, O. Kamal [EMAIL PROTECTED] wrote:
I am having 2 more issues, when starting asterisk I got the below message:
Dec 17 22:27:54 NOTICE[4554]: indications.c:505
ast_unregister_indication_country: Removed default indication country 'us'
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
signalling
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
answeronpolarityswitch
Dec 17 22:27:54 WARNING[4554]: chan_zap.c:10874 setup_zap: Ignoring
hanguponpolarityswitch
my setup is : softphone---softswitch(asterisk)Termination GW(asterisk
with TDM card)
when dialing from my softphone I got :
-- Executing Dial(SIP/10.8.0.6-0947d008, (Zap/g1/6159587)) in new
stack
Dec 17 23:04:16 WARNING[5052]: channel.c:2597 ast_request: No channel type
registered for '(Zap'
Dec 17 23:04:16 NOTICE[5052]: app_dial.c:1056 dial_exec_full: Unable to
create channel of type '(Zap' (cause 66 - Channel not implemented)
my extensions.conf file has:
[globals]
TRUNK=Zap/g1
[topstn]
exten=_2.,1,Dial,(${TRUNK}/${EXTEN:3})
Please help ...
Thanks,
On 12/12/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Dec 11, 2006 at 08:08:18PM -0500, Time Bandit wrote:
[channels]
context=default
signalling=fxs_ls
;channel=1-16
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
;accountcode=lss0101
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
To the best of my knowledge, all the settings you put after defining
the channles (channel= line) are useless. You have to set all the
settings BEFORE you define the channels.
Should be. However in practice after the first reload all of them will
be applied (in this specific case).
/me points again to genzaptelconf that should have made this thread
unnecessary.
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto: [EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] TDM2400
here is the latest update: in zaptel.conf i used fxsks=1-4 fxsks=5-8 fxsks=9-12 fxsks=13-16 zttool shows hardware OK ztcfg worked normally in zapata.conf when i define the channels channel=1-16 and restaring asterisk it gives the below errors: Dec 12 00:48:28 WARNING[3141]: chan_zap.c:921 zt_open: Unable to specify channel 1: No such device Dec 12 00:48:28 ERROR[3141]: chan_zap.c:6879 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Dec 12 00:48:28 ERROR[3141]: chan_zap.c:10307 setup_zap: Unable to register channel '1-16' Dec 12 00:48:28 WARNING[3141]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Dec 12 00:48:28 WARNING[3141]: loader.c:554 load_modules: Loading module chan_zap.so failed! When I remove the channel=1-16, it loads normally. zapata.conf is below: [channels] context=default signalling=fxs_ls ;channel=1-16 usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 ;accountcode=lss0101 answeronpolarityswitch=yes hanguponpolarityswitch=yes any clue? On 12/11/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 11, 2006 at 09:40:18AM +1100, Howard Lowndes wrote: O.Kamal wrote: I have 16 channels FXO (4 FXO Modules), I did follow the below link, but maybe I understand it wrong (what is a module and slot?), I need an example. http://kb.digium.com/entry/1/90/ http://kb.digium.com/entry/1/90/ For each FXO module, you should have a coresponding line that reads: fxs followed by the type of signalling (gs, ls, or ks) and the equals sign (=) followed by the position of the module times 4 minus 3 a dash, and then the number of the slot times 4. For example, if you had a FXO module on slot 2 of the board using loopstart signalling, the line would read: fxols=5-8, or if the module was on slot 5, the line would read: fxols=17-20 OK, try either: fxsks=1-16 or: fxsks=1-4 fxsks=5-8 fxsks=9-12 fxsks=13-16 probably the latter will be correct Both are. Alternatively, genzaptelconf (xpp/utils/genzaptelconf) will generate a working (though a bit verbose) configuration. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] TDM2400
I figured out the problem, it is the location of FXO boards on cards, channels are from 9-24 not 1-16. Thanks all for your help, specially Tzafrir, genzaptelconf shows it clearly. On 12/11/06, O. Kamal [EMAIL PROTECTED] wrote: here is the latest update: in zaptel.conf i used fxsks=1-4 fxsks=5-8 fxsks=9-12 fxsks=13-16 zttool shows hardware OK ztcfg worked normally in zapata.conf when i define the channels channel=1-16 and restaring asterisk it gives the below errors: Dec 12 00:48:28 WARNING[3141]: chan_zap.c:921 zt_open: Unable to specify channel 1: No such device Dec 12 00:48:28 ERROR[3141]: chan_zap.c:6879 mkintf: Unable to open channel 1: No such device here = 0, tmp-channel = 1, channel = 1 Dec 12 00:48:28 ERROR[3141]: chan_zap.c:10307 setup_zap: Unable to register channel '1-16' Dec 12 00:48:28 WARNING[3141]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Dec 12 00:48:28 WARNING[3141]: loader.c:554 load_modules: Loading module chan_zap.so failed! When I remove the channel=1-16, it loads normally. zapata.conf is below: [channels] context=default signalling=fxs_ls ;channel=1-16 usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 ;accountcode=lss0101 answeronpolarityswitch=yes hanguponpolarityswitch=yes any clue? On 12/11/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 11, 2006 at 09:40:18AM +1100, Howard Lowndes wrote: O.Kamal wrote: I have 16 channels FXO (4 FXO Modules), I did follow the below link, but maybe I understand it wrong (what is a module and slot?), I need an example. http://kb.digium.com/entry/1/90/ http://kb.digium.com/entry/1/90/ For each FXO module, you should have a coresponding line that reads: fxs followed by the type of signalling (gs, ls, or ks) and the equals sign (=) followed by the position of the module times 4 minus 3 a dash, and then the number of the slot times 4. For example, if you had a FXO module on slot 2 of the board using loopstart signalling, the line would read: fxols=5-8, or if the module was on slot 5, the line would read: fxols=17-20 OK, try either: fxsks=1-16 or: fxsks=1-4 fxsks=5-8 fxsks=9-12 fxsks=13-16 probably the latter will be correct Both are. Alternatively, genzaptelconf (xpp/utils/genzaptelconf) will generate a working (though a bit verbose) configuration. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] TDM2400
Yes, signalling should be fks_ks, channel line must be the last one, I will start live testing next week On 12/12/06, Time Bandit [EMAIL PROTECTED] wrote: [channels] context=default signalling=fxs_ls ;channel=1-16 usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 ;accountcode=lss0101 answeronpolarityswitch=yes hanguponpolarityswitch=yes To the best of my knowledge, all the settings you put after defining the channles (channel= line) are useless. You have to set all the settings BEFORE you define the channels. hth ___ --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
